[Aug 10 15:41:57] DEBUG[17897] rtp_engine.c: Destroyed RTP instance '0x1c51c018' [Aug 10 15:41:57] DEBUG[17874] devicestate.c: No provider found, checking channel drivers for SIP - 2219 [Aug 10 15:41:57] DEBUG[17874] chan_sip.c: Checking device state for peer 2219 [Aug 10 15:41:57] DEBUG[17874] devicestate.c: Changing state for SIP/2219 - state 1 (Not in use) [Aug 10 15:41:57] DEBUG[17874] devicestate.c: device 'SIP/2219' state '1' [Aug 10 15:41:57] DEBUG[17907] app_queue.c: Device 'SIP/2219' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Aug 10 15:42:00] DEBUG[17897] chan_sip.c: Auto destroying SIP dialog '3c267281d7fd-i8hj0yhc8i67' [Aug 10 15:42:00] DEBUG[17897] chan_sip.c: Destroying SIP dialog 3c267281d7fd-i8hj0yhc8i67 [Aug 10 15:42:00] VERBOSE[17897] chan_sip.c: Really destroying SIP dialog '3c267281d7fd-i8hj0yhc8i67' Method: BYE [Aug 10 15:42:00] DEBUG[17897] rtp_engine.c: Destroyed RTP instance '0x1c613e28' [Aug 10 15:42:16] VERBOSE[17897] chan_sip.c: <--- SIP read from UDP:192.168.2.210:2048 ---> INVITE sip:2212@192.168.1.84 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.210:2048;branch=z9hG4bK-s1jumb7mh8t9;rport From: "2210" ;tag=llqi1mlu1l To: Call-ID: 3c2672a5cbc0-o7llh150d2xu CSeq: 1 INVITE Max-Forwards: 70 Contact: ;reg-id=1 X-Serialnumber: 0004132500A7 P-Key-Flags: keys="3" User-Agent: snom300/8.4.32 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: 100rel, replaces, from-change Content-Type: application/sdp Content-Length: 349 v=0 o=root 2048061303 2048061303 IN IP4 192.168.2.210 s=call c=IN IP4 192.168.2.210 t=0 0 m=audio 17574 RTP/SAVP 8 0 101 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:i99dpczMNuFrrIJkKr09+4nfcuXTzsz5W11qjb74 a=direction:both a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <-------------> [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Header 0 [ 36]: INVITE sip:2212@192.168.1.84 SIP/2.0 [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Header 1 [ 69]: Via: SIP/2.0/UDP 192.168.2.210:2048;branch=z9hG4bK-s1jumb7mh8t9;rport [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Header 2 [ 51]: From: "2210" ;tag=llqi1mlu1l [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Header 3 [ 27]: To: [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Header 4 [ 34]: Call-ID: 3c2672a5cbc0-o7llh150d2xu [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Header 5 [ 14]: CSeq: 1 INVITE [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Header 7 [ 47]: Contact: ;reg-id=1 [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Header 8 [ 28]: X-Serialnumber: 0004132500A7 [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Header 9 [ 21]: P-Key-Flags: keys="3" [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Header 10 [ 26]: User-Agent: snom300/8.4.32 [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Header 11 [ 23]: Accept: application/sdp [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Header 12 [ 96]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Header 13 [ 42]: Allow-Events: talk, hold, refer, call-info [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Header 14 [ 40]: Supported: 100rel, replaces, from-change [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Header 15 [ 29]: Content-Type: application/sdp [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Header 16 [ 19]: Content-Length: 349 [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Header 17 [ 0]: [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Body 0 [ 3]: v=0 [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Body 1 [ 49]: o=root 2048061303 2048061303 IN IP4 192.168.2.210 [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Body 2 [ 6]: s=call [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Body 3 [ 22]: c=IN IP4 192.168.2.210 [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Body 5 [ 30]: m=audio 17574 RTP/SAVP 8 0 101 [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Body 6 [ 82]: a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:i99dpczMNuFrrIJkKr09+4nfcuXTzsz5W11qjb74 [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Body 7 [ 16]: a=direction:both [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Body 8 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Body 9 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Body 10 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Body 11 [ 15]: a=fmtp:101 0-16 [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Body 12 [ 10]: a=ptime:20 [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Body 13 [ 10]: a=sendrecv [Aug 10 15:42:16] VERBOSE[17897] chan_sip.c: --- (17 headers 14 lines) --- [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: = Looking for Call ID: 3c2672a5cbc0-o7llh150d2xu (Checking From) --From tag llqi1mlu1l --To-tag [Aug 10 15:42:16] DEBUG[17897] acl.c: For destination '192.168.2.210', our source address is '192.168.1.84'. [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.1.84:5060 [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Allocating new SIP dialog for 3c2672a5cbc0-o7llh150d2xu - INVITE (No RTP) [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Aug 10 15:42:16] DEBUG[17897] sip/reqresp_parser.c: Begin: parsing SIP "Supported: 100rel, replaces, from-change" [Aug 10 15:42:16] DEBUG[17897] sip/reqresp_parser.c: Found SIP option: -100rel- [Aug 10 15:42:16] DEBUG[17897] sip/reqresp_parser.c: Matched SIP option: 100rel [Aug 10 15:42:16] DEBUG[17897] sip/reqresp_parser.c: Found SIP option: -replaces- [Aug 10 15:42:16] DEBUG[17897] sip/reqresp_parser.c: Matched SIP option: replaces [Aug 10 15:42:16] DEBUG[17897] sip/reqresp_parser.c: Found SIP option: -from-change- [Aug 10 15:42:16] DEBUG[17897] sip/reqresp_parser.c: Matched SIP option: from-change [Aug 10 15:42:16] DEBUG[17897] netsock2.c: Splitting '192.168.2.210:2048' into... [Aug 10 15:42:16] DEBUG[17897] netsock2.c: ...host '192.168.2.210' and port '2048'. [Aug 10 15:42:16] VERBOSE[17897] chan_sip.c: Sending to 192.168.2.210:2048 (NAT) [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Initializing initreq for method INVITE - callid 3c2672a5cbc0-o7llh150d2xu [Aug 10 15:42:16] VERBOSE[17897] chan_sip.c: Using INVITE request as basis request - 3c2672a5cbc0-o7llh150d2xu [Aug 10 15:42:16] DEBUG[17897] netsock2.c: Splitting '192.168.1.84' into... [Aug 10 15:42:16] DEBUG[17897] netsock2.c: ...host '192.168.1.84' and port ''. [Aug 10 15:42:16] VERBOSE[17897] chan_sip.c: Found peer '2210' for '2210' from 192.168.2.210:2048 [Aug 10 15:42:16] VERBOSE[17897] chan_sip.c: <--- Reliably Transmitting (NAT) to 192.168.2.210:2048 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.2.210:2048;branch=z9hG4bK-s1jumb7mh8t9;received=192.168.2.210;rport=2048 From: "2210" ;tag=llqi1mlu1l To: ;tag=as36625af9 Call-ID: 3c2672a5cbc0-o7llh150d2xu CSeq: 1 INVITE Server: Asterisk PBX 1.8.15.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3456537b" Content-Length: 0 <------------> [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #226 [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 192.168.2.210:2048 [Aug 10 15:42:16] VERBOSE[17897] chan_sip.c: Scheduling destruction of SIP dialog '3c2672a5cbc0-o7llh150d2xu' in 6400 ms (Method: INVITE) [Aug 10 15:42:16] VERBOSE[17897] chan_sip.c: <--- SIP read from UDP:192.168.2.210:2048 ---> ACK sip:2212@192.168.1.84 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.210:2048;branch=z9hG4bK-s1jumb7mh8t9;rport From: "2210" ;tag=llqi1mlu1l To: ;tag=as36625af9 Call-ID: 3c2672a5cbc0-o7llh150d2xu CSeq: 1 ACK Max-Forwards: 70 Contact: ;reg-id=1 Content-Length: 0 <-------------> [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Header 0 [ 33]: ACK sip:2212@192.168.1.84 SIP/2.0 [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Header 1 [ 69]: Via: SIP/2.0/UDP 192.168.2.210:2048;branch=z9hG4bK-s1jumb7mh8t9;rport [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Header 2 [ 51]: From: "2210" ;tag=llqi1mlu1l [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Header 3 [ 42]: To: ;tag=as36625af9 [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Header 4 [ 34]: Call-ID: 3c2672a5cbc0-o7llh150d2xu [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Header 5 [ 11]: CSeq: 1 ACK [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Header 7 [ 47]: Contact: ;reg-id=1 [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Header 8 [ 17]: Content-Length: 0 [Aug 10 15:42:16] VERBOSE[17897] chan_sip.c: --- (9 headers 0 lines) --- [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: = Looking for Call ID: 3c2672a5cbc0-o7llh150d2xu (Checking From) --From tag llqi1mlu1l --To-tag as36625af9 [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #226 [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Stopping retransmission on '3c2672a5cbc0-o7llh150d2xu' of Response 1: Match Found [Aug 10 15:42:16] VERBOSE[17897] chan_sip.c: <--- SIP read from UDP:192.168.2.210:2048 ---> INVITE sip:2212@192.168.1.84 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.210:2048;branch=z9hG4bK-1mlusy0c7p2y;rport From: "2210" ;tag=llqi1mlu1l To: Call-ID: 3c2672a5cbc0-o7llh150d2xu CSeq: 2 INVITE Max-Forwards: 70 Contact: ;reg-id=1 X-Serialnumber: 0004132500A7 P-Key-Flags: keys="3" User-Agent: snom300/8.4.32 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: 100rel, replaces, from-change Authorization: Digest username="2210",realm="asterisk",nonce="3456537b",uri="sip:2212@192.168.1.84",response="939e5a923acb8d518968adb5287f818b",algorithm=MD5 Content-Type: application/sdp Content-Length: 349 v=0 o=root 2048061303 2048061303 IN IP4 192.168.2.210 s=call c=IN IP4 192.168.2.210 t=0 0 m=audio 17574 RTP/SAVP 8 0 101 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:i99dpczMNuFrrIJkKr09+4nfcuXTzsz5W11qjb74 a=direction:both a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <-------------> [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Header 0 [ 36]: INVITE sip:2212@192.168.1.84 SIP/2.0 [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Header 1 [ 69]: Via: SIP/2.0/UDP 192.168.2.210:2048;branch=z9hG4bK-1mlusy0c7p2y;rport [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Header 2 [ 51]: From: "2210" ;tag=llqi1mlu1l [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Header 3 [ 27]: To: [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Header 4 [ 34]: Call-ID: 3c2672a5cbc0-o7llh150d2xu [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Header 5 [ 14]: CSeq: 2 INVITE [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Header 7 [ 47]: Contact: ;reg-id=1 [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Header 8 [ 28]: X-Serialnumber: 0004132500A7 [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Header 9 [ 21]: P-Key-Flags: keys="3" [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Header 10 [ 26]: User-Agent: snom300/8.4.32 [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Header 11 [ 23]: Accept: application/sdp [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Header 12 [ 96]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Header 13 [ 42]: Allow-Events: talk, hold, refer, call-info [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Header 14 [ 40]: Supported: 100rel, replaces, from-change [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Header 15 [157]: Authorization: Digest username="2210",realm="asterisk",nonce="3456537b",uri="sip:2212@192.168.1.84",response="939e5a923acb8d518968adb5287f818b",algorithm=MD5 [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Header 16 [ 29]: Content-Type: application/sdp [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Header 17 [ 19]: Content-Length: 349 [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Header 18 [ 0]: [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Body 0 [ 3]: v=0 [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Body 1 [ 49]: o=root 2048061303 2048061303 IN IP4 192.168.2.210 [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Body 2 [ 6]: s=call [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Body 3 [ 22]: c=IN IP4 192.168.2.210 [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Body 5 [ 30]: m=audio 17574 RTP/SAVP 8 0 101 [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Body 6 [ 82]: a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:i99dpczMNuFrrIJkKr09+4nfcuXTzsz5W11qjb74 [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Body 7 [ 16]: a=direction:both [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Body 8 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Body 9 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Body 10 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Body 11 [ 15]: a=fmtp:101 0-16 [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Body 12 [ 10]: a=ptime:20 [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Body 13 [ 10]: a=sendrecv [Aug 10 15:42:16] VERBOSE[17897] chan_sip.c: --- (18 headers 14 lines) --- [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: = Looking for Call ID: 3c2672a5cbc0-o7llh150d2xu (Checking From) --From tag llqi1mlu1l --To-tag [Aug 10 15:42:16] DEBUG[17897] netsock2.c: Splitting '192.168.1.84' into... [Aug 10 15:42:16] DEBUG[17897] netsock2.c: ...host '192.168.1.84' and port ''. [Aug 10 15:42:16] DEBUG[17897] netsock2.c: Splitting '192.168.1.84' into... [Aug 10 15:42:16] DEBUG[17897] netsock2.c: ...host '192.168.1.84' and port ''. [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Aug 10 15:42:16] DEBUG[17897] netsock2.c: Splitting '192.168.2.210:2048' into... [Aug 10 15:42:16] DEBUG[17897] netsock2.c: ...host '192.168.2.210' and port '2048'. [Aug 10 15:42:16] VERBOSE[17897] chan_sip.c: Sending to 192.168.2.210:2048 (NAT) [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Initializing initreq for method INVITE - callid 3c2672a5cbc0-o7llh150d2xu [Aug 10 15:42:16] VERBOSE[17897] chan_sip.c: Using INVITE request as basis request - 3c2672a5cbc0-o7llh150d2xu [Aug 10 15:42:16] DEBUG[17897] netsock2.c: Splitting '192.168.1.84' into... [Aug 10 15:42:16] DEBUG[17897] netsock2.c: ...host '192.168.1.84' and port ''. [Aug 10 15:42:16] VERBOSE[17897] chan_sip.c: Found peer '2210' for '2210' from 192.168.2.210:2048 [Aug 10 15:42:16] DEBUG[17897] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x1c613e28' [Aug 10 15:42:16] DEBUG[17897] res_rtp_asterisk.c: Allocated port 13020 for RTP instance '0x1c613e28' [Aug 10 15:42:16] DEBUG[17897] rtp_engine.c: RTP instance '0x1c613e28' is setup and ready to go [Aug 10 15:42:16] DEBUG[17897] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x1c613e28' [Aug 10 15:42:16] VERBOSE[17897] netsock2.c: == Using SIP RTP CoS mark 5 [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Setting NAT on RTP to On [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Processing session-level SDP o=root 2048061303 2048061303 IN IP4 192.168.2.210... UNSUPPORTED OR FAILED. [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Processing session-level SDP s=call... UNSUPPORTED OR FAILED. [Aug 10 15:42:16] DEBUG[17897] netsock2.c: Splitting '192.168.2.210' into... [Aug 10 15:42:16] DEBUG[17897] netsock2.c: ...host '192.168.2.210' and port ''. [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.2.210... OK. [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Aug 10 15:42:16] VERBOSE[17897] chan_sip.c: Found RTP audio format 8 [Aug 10 15:42:16] DEBUG[17897] rtp_engine.c: Setting payload 8 based on m type on 0x416965b0 [Aug 10 15:42:16] VERBOSE[17897] chan_sip.c: Found RTP audio format 0 [Aug 10 15:42:16] DEBUG[17897] rtp_engine.c: Setting payload 0 based on m type on 0x416965b0 [Aug 10 15:42:16] VERBOSE[17897] chan_sip.c: Found RTP audio format 101 [Aug 10 15:42:16] DEBUG[17897] rtp_engine.c: Setting payload 101 based on m type on 0x416965b0 [Aug 10 15:42:16] DEBUG[17897] sip/sdp_crypto.c: local_key64 9LyhOHooXaHyxyek3sEe2mCxoG81foVM00WoOcci len 40 [Aug 10 15:42:16] DEBUG[17897] res_srtp.c: Adding new policy for SSRC 416744333 [Aug 10 15:42:16] DEBUG[17897] sip/sdp_crypto.c: SRTP policy activated [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Processing media-level (audio) SDP a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:i99dpczMNuFrrIJkKr09+4nfcuXTzsz5W11qjb74... OK. [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Processing media-level (audio) SDP a=direction:both... UNSUPPORTED OR FAILED. [Aug 10 15:42:16] VERBOSE[17897] chan_sip.c: Found audio description format PCMA for ID 8 [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Aug 10 15:42:16] VERBOSE[17897] chan_sip.c: Found audio description format PCMU for ID 0 [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Aug 10 15:42:16] VERBOSE[17897] chan_sip.c: Found audio description format telephone-event for ID 101 [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED OR FAILED. [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Aug 10 15:42:16] DEBUG[17897] rtp_engine.c: Incorporating payload 0 on 0x416965b0 [Aug 10 15:42:16] DEBUG[17897] rtp_engine.c: Incorporating payload 8 on 0x416965b0 [Aug 10 15:42:16] DEBUG[17897] rtp_engine.c: Incorporating payload 101 on 0x416965b0 [Aug 10 15:42:16] VERBOSE[17897] chan_sip.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) [Aug 10 15:42:16] VERBOSE[17897] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Aug 10 15:42:16] DEBUG[17897] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x1c613e28' [Aug 10 15:42:16] VERBOSE[17897] chan_sip.c: Peer audio RTP is at port 192.168.2.210:17574 [Aug 10 15:42:16] DEBUG[17897] rtp_engine.c: Copying payload 0 from 0x416965b0 to 0x1c613ff0 [Aug 10 15:42:16] DEBUG[17897] rtp_engine.c: Copying payload 8 from 0x416965b0 to 0x1c613ff0 [Aug 10 15:42:16] DEBUG[17897] rtp_engine.c: Copying payload 101 from 0x416965b0 to 0x1c613ff0 [Aug 10 15:42:16] DEBUG[17897] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0x1c613e28' [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Checking SIP call limits for device 2210 [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Updating call counter for incoming call [Aug 10 15:42:16] DEBUG[17897] netsock2.c: Splitting '192.168.1.84' into... [Aug 10 15:42:16] DEBUG[17897] netsock2.c: ...host '192.168.1.84' and port ''. [Aug 10 15:42:16] DEBUG[17897] netsock2.c: Splitting '192.168.1.84' into... [Aug 10 15:42:16] DEBUG[17897] netsock2.c: ...host '192.168.1.84' and port ''. [Aug 10 15:42:16] VERBOSE[17897] chan_sip.c: Looking for 2212 in test (domain 192.168.1.84) [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: *** Our native formats are 0x8 (alaw) [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: *** Joint capabilities are 0xc (ulaw|alaw) [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: *** Our capabilities are 0xc (ulaw|alaw) [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x8 (alaw) [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: This channel will not be able to handle video. [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: build_route: Contact hop: ;reg-id=1 [Aug 10 15:42:16] VERBOSE[17897] chan_sip.c: list_route: hop: [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: SIP/2210-0000000b: New call is still down.... Trying... [Aug 10 15:42:16] VERBOSE[17897] chan_sip.c: <--- Transmitting (NAT) to 192.168.2.210:2048 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.2.210:2048;branch=z9hG4bK-1mlusy0c7p2y;received=192.168.2.210;rport=2048 From: "2210" ;tag=llqi1mlu1l To: Call-ID: 3c2672a5cbc0-o7llh150d2xu CSeq: 2 INVITE Server: Asterisk PBX 1.8.15.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 192.168.2.210:2048 [Aug 10 15:42:16] DEBUG[17874] devicestate.c: No provider found, checking channel drivers for SIP - 2210 [Aug 10 15:42:16] DEBUG[17874] chan_sip.c: Checking device state for peer 2210 [Aug 10 15:42:16] DEBUG[17874] devicestate.c: Changing state for SIP/2210 - state 1 (Not in use) [Aug 10 15:42:16] DEBUG[17874] devicestate.c: device 'SIP/2210' state '1' [Aug 10 15:42:16] DEBUG[17907] app_queue.c: Device 'SIP/2210' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Aug 10 15:42:16] DEBUG[18238] pbx.c: Result of 'EXTEN' is '2212' [Aug 10 15:42:16] DEBUG[18238] pbx.c: Launching 'Dial' [Aug 10 15:42:16] VERBOSE[18238] pbx.c: -- Executing [2212@test:1] Dial("SIP/2210-0000000b", "SIP/2212") in new stack [Aug 10 15:42:16] DEBUG[18238] chan_sip.c: Asked to create a SIP channel with formats: 0x8 (alaw) [Aug 10 15:42:16] DEBUG[18238] chan_sip.c: Allocating new SIP dialog for 688a9c5277171344637477bc6044c6a5@127.0.0.1:5060 - INVITE (No RTP) [Aug 10 15:42:16] DEBUG[18238] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x1c51c018' [Aug 10 15:42:16] DEBUG[18238] res_rtp_asterisk.c: Allocated port 11504 for RTP instance '0x1c51c018' [Aug 10 15:42:16] DEBUG[18238] rtp_engine.c: RTP instance '0x1c51c018' is setup and ready to go [Aug 10 15:42:16] DEBUG[18238] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x1c51c018' [Aug 10 15:42:16] VERBOSE[18238] netsock2.c: == Using SIP RTP CoS mark 5 [Aug 10 15:42:16] DEBUG[18238] chan_sip.c: Setting NAT on RTP to On [Aug 10 15:42:16] DEBUG[18238] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 10 15:42:16] DEBUG[18238] acl.c: For destination '192.168.1.102', our source address is '192.168.1.84'. [Aug 10 15:42:16] DEBUG[18238] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.1.84:5060 [Aug 10 15:42:16] DEBUG[18238] chan_sip.c: *** Our native formats are 0x8 (alaw) [Aug 10 15:42:16] DEBUG[18238] chan_sip.c: *** Joint capabilities are 0x8 (alaw) [Aug 10 15:42:16] DEBUG[18238] chan_sip.c: *** Our capabilities are 0xc (ulaw|alaw) [Aug 10 15:42:16] DEBUG[18238] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x8 (alaw) [Aug 10 15:42:16] DEBUG[18238] chan_sip.c: *** Our preferred formats from the incoming channel are 0x8 (alaw) [Aug 10 15:42:16] DEBUG[18238] chan_sip.c: This channel will not be able to handle video. [Aug 10 15:42:16] DEBUG[18238] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 10 15:42:16] DEBUG[18238] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 10 15:42:16] DEBUG[18238] rtp_engine.c: Seeded SDP of 'SIP/2212-0000000c' with that of 'SIP/2210-0000000b' [Aug 10 15:42:16] DEBUG[18238] channel.c: Not copying variable DIALEDTIME. [Aug 10 15:42:16] DEBUG[18238] channel.c: Not copying variable ANSWEREDTIME. [Aug 10 15:42:16] DEBUG[18238] channel.c: Not copying variable DIALEDPEERNAME. [Aug 10 15:42:16] DEBUG[18238] channel.c: Not copying variable DIALEDPEERNUMBER. [Aug 10 15:42:16] DEBUG[18238] channel.c: Not copying variable DIALSTATUS. [Aug 10 15:42:16] DEBUG[18238] channel.c: Not copying variable SIPCALLID. [Aug 10 15:42:16] DEBUG[18238] channel.c: Not copying variable SIPDOMAIN. [Aug 10 15:42:16] DEBUG[18238] channel.c: Not copying variable SIPURI. [Aug 10 15:42:16] DEBUG[18238] chan_sip.c: Direct media not possible when using SRTP, ignoring canreinvite setting [Aug 10 15:42:16] DEBUG[18238] chan_sip.c: Outgoing Call for 2212 [Aug 10 15:42:16] DEBUG[18238] chan_sip.c: Updating call counter for outgoing call [Aug 10 15:42:16] DEBUG[18238] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: False Text flag: False [Aug 10 15:42:16] DEBUG[18238] chan_sip.c: ** Our prefcodec: 0x8 (alaw) [Aug 10 15:42:16] VERBOSE[18238] chan_sip.c: Audio is at 11504 [Aug 10 15:42:16] DEBUG[18238] sip/sdp_crypto.c: local_key64 MpH0NOA3Z4kUPjpwr3NHl/6C20uTb1hsmwyjzTSZ len 40 [Aug 10 15:42:16] VERBOSE[18238] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Aug 10 15:42:16] VERBOSE[18238] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Aug 10 15:42:16] VERBOSE[18238] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 10 15:42:16] DEBUG[18238] chan_sip.c: -- Done with adding codecs to SDP [Aug 10 15:42:16] DEBUG[18238] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Aug 10 15:42:16] DEBUG[18238] chan_sip.c: Initializing initreq for method INVITE - callid 542a95eb6183b28264e5cfc965b562ff@192.168.1.84:5060 [Aug 10 15:42:16] DEBUG[18238] chan_sip.c: Header 0 [ 56]: INVITE sip:2212@192.168.1.102:2048;line=z1zv7jsj SIP/2.0 [Aug 10 15:42:16] DEBUG[18238] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK38115ba4;rport [Aug 10 15:42:16] DEBUG[18238] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 10 15:42:16] DEBUG[18238] chan_sip.c: Header 3 [ 54]: From: "Unknown" ;tag=as497f3fae [Aug 10 15:42:16] DEBUG[18238] chan_sip.c: Header 4 [ 47]: To: [Aug 10 15:42:16] DEBUG[18238] chan_sip.c: Header 5 [ 37]: Contact: [Aug 10 15:42:16] DEBUG[18238] chan_sip.c: Header 6 [ 59]: Call-ID: 542a95eb6183b28264e5cfc965b562ff@192.168.1.84:5060 [Aug 10 15:42:16] DEBUG[18238] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 10 15:42:16] DEBUG[18238] chan_sip.c: Header 8 [ 33]: User-Agent: Asterisk PBX 1.8.15.0 [Aug 10 15:42:16] DEBUG[18238] chan_sip.c: Header 9 [ 35]: Date: Fri, 10 Aug 2012 13:42:16 GMT [Aug 10 15:42:16] DEBUG[18238] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 10 15:42:16] DEBUG[18238] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 10 15:42:16] DEBUG[18238] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 10 15:42:16] VERBOSE[18238] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.102:2048: INVITE sip:2212@192.168.1.102:2048;line=z1zv7jsj SIP/2.0 Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK38115ba4;rport Max-Forwards: 70 From: "Unknown" ;tag=as497f3fae To: Contact: Call-ID: 542a95eb6183b28264e5cfc965b562ff@192.168.1.84:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.15.0 Date: Fri, 10 Aug 2012 13:42:16 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 373 v=0 o=root 1332746239 1332746239 IN IP4 192.168.1.84 s=Asterisk PBX 1.8.15.0 c=IN IP4 192.168.1.84 t=0 0 m=audio 11504 RTP/SAVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:MpH0NOA3Z4kUPjpwr3NHl/6C20uTb1hsmwyjzTSZ --- [Aug 10 15:42:16] DEBUG[18238] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #229 [Aug 10 15:42:16] DEBUG[18238] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.1.102:2048 [Aug 10 15:42:16] VERBOSE[18238] app_dial.c: -- Called SIP/2212 [Aug 10 15:42:16] VERBOSE[17897] chan_sip.c: <--- SIP read from UDP:192.168.1.102:2048 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK38115ba4;rport=5060 From: "Unknown" ;tag=as497f3fae To: ;tag=qya7y2dy4z Call-ID: 542a95eb6183b28264e5cfc965b562ff@192.168.1.84:5060 CSeq: 102 INVITE Contact: ;reg-id=1 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Content-Length: 0 <-------------> [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Header 0 [ 19]: SIP/2.0 180 Ringing [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Header 1 [ 68]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK38115ba4;rport=5060 [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Header 2 [ 54]: From: "Unknown" ;tag=as497f3fae [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Header 3 [ 62]: To: ;tag=qya7y2dy4z [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Header 4 [ 59]: Call-ID: 542a95eb6183b28264e5cfc965b562ff@192.168.1.84:5060 [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Header 6 [ 61]: Contact: ;reg-id=1 [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Header 7 [ 96]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Header 8 [ 42]: Allow-Events: talk, hold, refer, call-info [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Aug 10 15:42:16] VERBOSE[17897] chan_sip.c: --- (10 headers 0 lines) --- [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: = Looking for Call ID: 542a95eb6183b28264e5cfc965b562ff@192.168.1.84:5060 (Checking To) --From tag as497f3fae --To-tag qya7y2dy4z [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: *** SIP TIMER: Cancelling retransmission #229 - INVITE (got response) [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '542a95eb6183b28264e5cfc965b562ff@192.168.1.84:5060' Request 102: Found [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: SIP response 180 to standard invite [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: build_route: Contact hop: ;reg-id=1 [Aug 10 15:42:16] VERBOSE[17897] chan_sip.c: list_route: hop: [Aug 10 15:42:16] DEBUG[17874] devicestate.c: No provider found, checking channel drivers for SIP - 2212 [Aug 10 15:42:16] DEBUG[17874] chan_sip.c: Checking device state for peer 2212 [Aug 10 15:42:16] DEBUG[17874] devicestate.c: Changing state for SIP/2212 - state 1 (Not in use) [Aug 10 15:42:16] DEBUG[17874] devicestate.c: device 'SIP/2212' state '1' [Aug 10 15:42:16] DEBUG[17907] app_queue.c: Device 'SIP/2212' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Aug 10 15:42:16] VERBOSE[18238] app_dial.c: -- SIP/2212-0000000c is ringing [Aug 10 15:42:16] DEBUG[18238] rtp_engine.c: Setting early bridge SDP of 'SIP/2210-0000000b' with that of 'SIP/2212-0000000c' [Aug 10 15:42:16] VERBOSE[18238] chan_sip.c: <--- Transmitting (NAT) to 192.168.2.210:2048 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.2.210:2048;branch=z9hG4bK-1mlusy0c7p2y;received=192.168.2.210;rport=2048 From: "2210" ;tag=llqi1mlu1l To: ;tag=as77bb01b2 Call-ID: 3c2672a5cbc0-o7llh150d2xu CSeq: 2 INVITE Server: Asterisk PBX 1.8.15.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> [Aug 10 15:42:16] DEBUG[18238] chan_sip.c: Trying to put 'SIP/2.0 180' onto UDP socket destined for 192.168.2.210:2048 [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Auto destroying SIP dialog '50250e9b9684-8d3g9c7o4hy2' [Aug 10 15:42:16] DEBUG[17897] chan_sip.c: Destroying SIP dialog 50250e9b9684-8d3g9c7o4hy2 [Aug 10 15:42:16] VERBOSE[17897] chan_sip.c: Really destroying SIP dialog '50250e9b9684-8d3g9c7o4hy2' Method: REGISTER [Aug 10 15:42:17] VERBOSE[17897] chan_sip.c: <--- SIP read from UDP:192.168.1.102:2048 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK38115ba4;rport=5060 From: "Unknown" ;tag=as497f3fae To: ;tag=qya7y2dy4z Call-ID: 542a95eb6183b28264e5cfc965b562ff@192.168.1.84:5060 CSeq: 102 INVITE Contact: ;reg-id=1 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Content-Length: 0 <-------------> [Aug 10 15:42:17] DEBUG[17897] chan_sip.c: Header 0 [ 19]: SIP/2.0 180 Ringing [Aug 10 15:42:17] DEBUG[17897] chan_sip.c: Header 1 [ 68]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK38115ba4;rport=5060 [Aug 10 15:42:17] DEBUG[17897] chan_sip.c: Header 2 [ 54]: From: "Unknown" ;tag=as497f3fae [Aug 10 15:42:17] DEBUG[17897] chan_sip.c: Header 3 [ 62]: To: ;tag=qya7y2dy4z [Aug 10 15:42:17] DEBUG[17897] chan_sip.c: Header 4 [ 59]: Call-ID: 542a95eb6183b28264e5cfc965b562ff@192.168.1.84:5060 [Aug 10 15:42:17] DEBUG[17897] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 10 15:42:17] DEBUG[17897] chan_sip.c: Header 6 [ 61]: Contact: ;reg-id=1 [Aug 10 15:42:17] DEBUG[17897] chan_sip.c: Header 7 [ 96]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE [Aug 10 15:42:17] DEBUG[17897] chan_sip.c: Header 8 [ 42]: Allow-Events: talk, hold, refer, call-info [Aug 10 15:42:17] DEBUG[17897] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Aug 10 15:42:17] VERBOSE[17897] chan_sip.c: --- (10 headers 0 lines) --- [Aug 10 15:42:17] DEBUG[17897] chan_sip.c: = Looking for Call ID: 542a95eb6183b28264e5cfc965b562ff@192.168.1.84:5060 (Checking To) --From tag as497f3fae --To-tag qya7y2dy4z [Aug 10 15:42:17] DEBUG[17897] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '542a95eb6183b28264e5cfc965b562ff@192.168.1.84:5060' Request 102: Found [Aug 10 15:42:17] DEBUG[17897] chan_sip.c: SIP response 180 to standard invite [Aug 10 15:42:17] DEBUG[17897] chan_sip.c: build_route: Contact hop: ;reg-id=1 [Aug 10 15:42:17] VERBOSE[17897] chan_sip.c: list_route: hop: [Aug 10 15:42:17] VERBOSE[18238] app_dial.c: -- SIP/2212-0000000c is ringing [Aug 10 15:42:17] DEBUG[18238] rtp_engine.c: Setting early bridge SDP of 'SIP/2210-0000000b' with that of 'SIP/2212-0000000c' [Aug 10 15:42:18] DEBUG[18238] res_rtp_asterisk.c: RTCP NAT: Got RTCP from other end. Now sending to address 192.168.1.102:15779 [Aug 10 15:42:18] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 82 bytes [Aug 10 15:42:18] DEBUG[18238] res_rtp_asterisk.c: RTCP Read too short [Aug 10 15:42:18] DEBUG[18238] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x1c51c018' [Aug 10 15:42:18] VERBOSE[17897] chan_sip.c: <--- SIP read from UDP:192.168.1.102:2048 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK38115ba4;rport=5060 From: "Unknown" ;tag=as497f3fae To: ;tag=qya7y2dy4z Call-ID: 542a95eb6183b28264e5cfc965b562ff@192.168.1.84:5060 CSeq: 102 INVITE Contact: ;reg-id=1 User-Agent: snom300/8.4.32 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Content-Type: application/sdp Content-Length: 305 v=0 o=root 766647476 766647477 IN IP4 192.168.1.102 s=call c=IN IP4 192.168.1.102 t=0 0 m=audio 15778 RTP/SAVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:qbgnnE+yg912UrjXEFEB9xGhUAFWFmD4Cd7u3jyG a=sendrecv <-------------> [Aug 10 15:42:18] DEBUG[17897] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 Ok [Aug 10 15:42:18] DEBUG[17897] chan_sip.c: Header 1 [ 68]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK38115ba4;rport=5060 [Aug 10 15:42:18] DEBUG[17897] chan_sip.c: Header 2 [ 54]: From: "Unknown" ;tag=as497f3fae [Aug 10 15:42:18] DEBUG[17897] chan_sip.c: Header 3 [ 62]: To: ;tag=qya7y2dy4z [Aug 10 15:42:18] DEBUG[17897] chan_sip.c: Header 4 [ 59]: Call-ID: 542a95eb6183b28264e5cfc965b562ff@192.168.1.84:5060 [Aug 10 15:42:18] DEBUG[17897] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 10 15:42:18] DEBUG[17897] chan_sip.c: Header 6 [ 61]: Contact: ;reg-id=1 [Aug 10 15:42:18] DEBUG[17897] chan_sip.c: Header 7 [ 26]: User-Agent: snom300/8.4.32 [Aug 10 15:42:18] DEBUG[17897] chan_sip.c: Header 8 [ 96]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE [Aug 10 15:42:18] DEBUG[17897] chan_sip.c: Header 9 [ 42]: Allow-Events: talk, hold, refer, call-info [Aug 10 15:42:18] DEBUG[17897] chan_sip.c: Header 10 [ 47]: Supported: timer, 100rel, replaces, from-change [Aug 10 15:42:18] DEBUG[17897] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Aug 10 15:42:18] DEBUG[17897] chan_sip.c: Header 12 [ 19]: Content-Length: 305 [Aug 10 15:42:18] DEBUG[17897] chan_sip.c: Header 13 [ 0]: [Aug 10 15:42:18] DEBUG[17897] chan_sip.c: Body 0 [ 3]: v=0 [Aug 10 15:42:18] DEBUG[17897] chan_sip.c: Body 1 [ 47]: o=root 766647476 766647477 IN IP4 192.168.1.102 [Aug 10 15:42:18] DEBUG[17897] chan_sip.c: Body 2 [ 6]: s=call [Aug 10 15:42:18] DEBUG[17897] chan_sip.c: Body 3 [ 22]: c=IN IP4 192.168.1.102 [Aug 10 15:42:18] DEBUG[17897] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 10 15:42:18] DEBUG[17897] chan_sip.c: Body 5 [ 28]: m=audio 15778 RTP/SAVP 8 101 [Aug 10 15:42:18] DEBUG[17897] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 10 15:42:18] DEBUG[17897] chan_sip.c: Body 7 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 10 15:42:18] DEBUG[17897] chan_sip.c: Body 8 [ 15]: a=fmtp:101 0-16 [Aug 10 15:42:18] DEBUG[17897] chan_sip.c: Body 9 [ 10]: a=ptime:20 [Aug 10 15:42:18] DEBUG[17897] chan_sip.c: Body 10 [ 82]: a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:qbgnnE+yg912UrjXEFEB9xGhUAFWFmD4Cd7u3jyG [Aug 10 15:42:18] DEBUG[17897] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 10 15:42:18] VERBOSE[17897] chan_sip.c: --- (13 headers 12 lines) --- [Aug 10 15:42:18] DEBUG[17897] chan_sip.c: = Looking for Call ID: 542a95eb6183b28264e5cfc965b562ff@192.168.1.84:5060 (Checking To) --From tag as497f3fae --To-tag qya7y2dy4z [Aug 10 15:42:18] DEBUG[17897] chan_sip.c: Acked pending invite 102 [Aug 10 15:42:18] DEBUG[17897] chan_sip.c: Stopping retransmission on '542a95eb6183b28264e5cfc965b562ff@192.168.1.84:5060' of Request 102: Match Found [Aug 10 15:42:18] DEBUG[17897] chan_sip.c: SIP response 200 to standard invite [Aug 10 15:42:18] DEBUG[17897] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Aug 10 15:42:18] DEBUG[17897] chan_sip.c: Processing session-level SDP o=root 766647476 766647477 IN IP4 192.168.1.102... UNSUPPORTED OR FAILED. [Aug 10 15:42:18] DEBUG[17897] chan_sip.c: Processing session-level SDP s=call... UNSUPPORTED OR FAILED. [Aug 10 15:42:18] DEBUG[17897] netsock2.c: Splitting '192.168.1.102' into... [Aug 10 15:42:18] DEBUG[17897] netsock2.c: ...host '192.168.1.102' and port ''. [Aug 10 15:42:18] DEBUG[17897] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.1.102... OK. [Aug 10 15:42:18] DEBUG[17897] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Aug 10 15:42:18] VERBOSE[17897] chan_sip.c: Found RTP audio format 8 [Aug 10 15:42:18] DEBUG[17897] rtp_engine.c: Setting payload 8 based on m type on 0x41695d00 [Aug 10 15:42:18] VERBOSE[17897] chan_sip.c: Found RTP audio format 101 [Aug 10 15:42:18] DEBUG[17897] rtp_engine.c: Setting payload 101 based on m type on 0x41695d00 [Aug 10 15:42:18] VERBOSE[17897] chan_sip.c: Found audio description format PCMA for ID 8 [Aug 10 15:42:18] DEBUG[17897] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Aug 10 15:42:18] VERBOSE[17897] chan_sip.c: Found audio description format telephone-event for ID 101 [Aug 10 15:42:18] DEBUG[17897] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Aug 10 15:42:18] DEBUG[17897] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED OR FAILED. [Aug 10 15:42:18] DEBUG[17897] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Aug 10 15:42:18] DEBUG[17897] res_srtp.c: Adding new policy for SSRC 647730817 [Aug 10 15:42:18] DEBUG[17897] sip/sdp_crypto.c: SRTP policy activated [Aug 10 15:42:18] DEBUG[17897] chan_sip.c: Processing media-level (audio) SDP a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:qbgnnE+yg912UrjXEFEB9xGhUAFWFmD4Cd7u3jyG... OK. [Aug 10 15:42:18] DEBUG[17897] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Aug 10 15:42:18] DEBUG[17897] rtp_engine.c: Incorporating payload 8 on 0x41695d00 [Aug 10 15:42:18] DEBUG[17897] rtp_engine.c: Incorporating payload 101 on 0x41695d00 [Aug 10 15:42:18] VERBOSE[17897] chan_sip.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) [Aug 10 15:42:18] VERBOSE[17897] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Aug 10 15:42:18] DEBUG[17897] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x1c51c018' [Aug 10 15:42:18] VERBOSE[17897] chan_sip.c: Peer audio RTP is at port 192.168.1.102:15778 [Aug 10 15:42:18] DEBUG[17897] rtp_engine.c: Copying payload 8 from 0x41695d00 to 0x1c51c1e0 [Aug 10 15:42:18] DEBUG[17897] rtp_engine.c: Copying payload 101 from 0x41695d00 to 0x1c51c1e0 [Aug 10 15:42:18] DEBUG[17897] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0x1c51c018' [Aug 10 15:42:18] DEBUG[17897] chan_sip.c: We're settling with these formats: 0x8 (alaw) [Aug 10 15:42:18] DEBUG[17897] chan_sip.c: We have an owner, now see if we need to change this call [Aug 10 15:42:18] DEBUG[17897] chan_sip.c: Updating call counter for outgoing call [Aug 10 15:42:18] DEBUG[17897] chan_sip.c: build_route: Contact hop: ;reg-id=1 [Aug 10 15:42:18] VERBOSE[17897] chan_sip.c: list_route: hop: [Aug 10 15:42:18] DEBUG[17897] chan_sip.c: Strict routing enforced for session 542a95eb6183b28264e5cfc965b562ff@192.168.1.84:5060 [Aug 10 15:42:18] VERBOSE[17897] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 10 15:42:18] DEBUG[17897] netsock2.c: Splitting '192.168.1.102:2048' into... [Aug 10 15:42:18] DEBUG[17897] netsock2.c: ...host '192.168.1.102' and port '2048'. [Aug 10 15:42:18] VERBOSE[17897] chan_sip.c: set_destination: set destination to 192.168.1.102:2048 [Aug 10 15:42:18] VERBOSE[17897] chan_sip.c: Transmitting (NAT) to 192.168.1.102:2048: ACK sip:2212@192.168.1.102:2048;line=z1zv7jsj SIP/2.0 Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK4e3646a2;rport Max-Forwards: 70 From: "Unknown" ;tag=as497f3fae To: ;tag=qya7y2dy4z Contact: Call-ID: 542a95eb6183b28264e5cfc965b562ff@192.168.1.84:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 1.8.15.0 Content-Length: 0 --- [Aug 10 15:42:18] DEBUG[17897] chan_sip.c: Trying to put 'ACK sip:221' onto UDP socket destined for 192.168.1.102:2048 [Aug 10 15:42:18] VERBOSE[18238] app_dial.c: -- SIP/2212-0000000c answered SIP/2210-0000000b [Aug 10 15:42:18] DEBUG[18238] rtp_engine.c: Setting early bridge SDP of 'SIP/2210-0000000b' with that of 'SIP/2212-0000000c' [Aug 10 15:42:18] DEBUG[18238] chan_sip.c: SIP answering channel: SIP/2210-0000000b [Aug 10 15:42:18] DEBUG[18238] res_rtp_asterisk.c: Setting the marker bit due to a source update [Aug 10 15:42:18] DEBUG[18238] chan_sip.c: Setting framing from config on incoming call [Aug 10 15:42:18] DEBUG[18238] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: True Text flag: True [Aug 10 15:42:18] DEBUG[18238] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Aug 10 15:42:18] VERBOSE[18238] chan_sip.c: Audio is at 13020 [Aug 10 15:42:18] VERBOSE[18238] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Aug 10 15:42:18] VERBOSE[18238] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Aug 10 15:42:18] VERBOSE[18238] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 10 15:42:18] DEBUG[18238] chan_sip.c: -- Done with adding codecs to SDP [Aug 10 15:42:18] DEBUG[18238] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Aug 10 15:42:18] VERBOSE[18238] chan_sip.c: <--- Reliably Transmitting (NAT) to 192.168.2.210:2048 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.210:2048;branch=z9hG4bK-1mlusy0c7p2y;received=192.168.2.210;rport=2048 From: "2210" ;tag=llqi1mlu1l To: ;tag=as77bb01b2 Call-ID: 3c2672a5cbc0-o7llh150d2xu CSeq: 2 INVITE Server: Asterisk PBX 1.8.15.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 371 v=0 o=root 750441157 750441157 IN IP4 192.168.1.84 s=Asterisk PBX 1.8.15.0 c=IN IP4 192.168.1.84 t=0 0 m=audio 13020 RTP/SAVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:9LyhOHooXaHyxyek3sEe2mCxoG81foVM00WoOcci <------------> [Aug 10 15:42:18] DEBUG[18238] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #232 [Aug 10 15:42:18] DEBUG[18238] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.2.210:2048 [Aug 10 15:42:18] DEBUG[18238] features.c: bridge answer set, chan answer set [Aug 10 15:42:18] DEBUG[18238] features.c: Removing dialed interfaces datastore on SIP/2212-0000000c since we're bridging [Aug 10 15:42:18] DEBUG[18238] res_rtp_asterisk.c: Setting the marker bit due to a source update [Aug 10 15:42:18] DEBUG[18238] res_rtp_asterisk.c: Setting the marker bit due to a source update [Aug 10 15:42:18] DEBUG[17874] devicestate.c: No provider found, checking channel drivers for SIP - 2212 [Aug 10 15:42:18] DEBUG[17874] chan_sip.c: Checking device state for peer 2212 [Aug 10 15:42:18] DEBUG[17874] devicestate.c: Changing state for SIP/2212 - state 1 (Not in use) [Aug 10 15:42:18] DEBUG[17874] devicestate.c: device 'SIP/2212' state '1' [Aug 10 15:42:18] DEBUG[17874] devicestate.c: No provider found, checking channel drivers for SIP - 2210 [Aug 10 15:42:18] DEBUG[17874] chan_sip.c: Checking device state for peer 2210 [Aug 10 15:42:18] DEBUG[17874] devicestate.c: Changing state for SIP/2210 - state 1 (Not in use) [Aug 10 15:42:18] DEBUG[17874] devicestate.c: device 'SIP/2210' state '1' [Aug 10 15:42:18] DEBUG[17907] app_queue.c: Device 'SIP/2212' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Aug 10 15:42:18] DEBUG[17907] app_queue.c: Device 'SIP/2210' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Aug 10 15:42:18] VERBOSE[17897] chan_sip.c: <--- SIP read from UDP:192.168.2.210:2048 ---> ACK sip:2212@192.168.1.84:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.210:2048;branch=z9hG4bK-r0jxtoqf84lj;rport From: "2210" ;tag=llqi1mlu1l To: ;tag=as77bb01b2 Call-ID: 3c2672a5cbc0-o7llh150d2xu CSeq: 2 ACK Max-Forwards: 70 Contact: ;reg-id=1 Content-Length: 0 <-------------> [Aug 10 15:42:18] DEBUG[17897] chan_sip.c: Header 0 [ 38]: ACK sip:2212@192.168.1.84:5060 SIP/2.0 [Aug 10 15:42:18] DEBUG[17897] chan_sip.c: Header 1 [ 69]: Via: SIP/2.0/UDP 192.168.2.210:2048;branch=z9hG4bK-r0jxtoqf84lj;rport [Aug 10 15:42:18] DEBUG[17897] chan_sip.c: Header 2 [ 51]: From: "2210" ;tag=llqi1mlu1l [Aug 10 15:42:18] DEBUG[17897] chan_sip.c: Header 3 [ 42]: To: ;tag=as77bb01b2 [Aug 10 15:42:18] DEBUG[17897] chan_sip.c: Header 4 [ 34]: Call-ID: 3c2672a5cbc0-o7llh150d2xu [Aug 10 15:42:18] DEBUG[17897] chan_sip.c: Header 5 [ 11]: CSeq: 2 ACK [Aug 10 15:42:18] DEBUG[17897] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Aug 10 15:42:18] DEBUG[17897] chan_sip.c: Header 7 [ 47]: Contact: ;reg-id=1 [Aug 10 15:42:18] DEBUG[17897] chan_sip.c: Header 8 [ 17]: Content-Length: 0 [Aug 10 15:42:18] VERBOSE[17897] chan_sip.c: --- (9 headers 0 lines) --- [Aug 10 15:42:18] DEBUG[17897] chan_sip.c: = Looking for Call ID: 3c2672a5cbc0-o7llh150d2xu (Checking From) --From tag llqi1mlu1l --To-tag as77bb01b2 [Aug 10 15:42:18] DEBUG[17897] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Aug 10 15:42:18] DEBUG[17897] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #232 [Aug 10 15:42:18] DEBUG[17897] chan_sip.c: Stopping retransmission on '3c2672a5cbc0-o7llh150d2xu' of Response 2: Match Found [Aug 10 15:42:19] DEBUG[18238] res_rtp_asterisk.c: Ooh, format changed from unknown to alaw [Aug 10 15:42:19] DEBUG[18238] res_rtp_asterisk.c: Created smoother: format: alaw ms: 20 len: 160 [Aug 10 15:42:19] DEBUG[18238] res_rtp_asterisk.c: Starting RTCP transmission on RTP instance '0x1c613e28' [Aug 10 15:42:19] DEBUG[18238] res_rtp_asterisk.c: Ooh, format changed from unknown to alaw [Aug 10 15:42:19] DEBUG[18238] res_rtp_asterisk.c: Created smoother: format: alaw ms: 20 len: 160 [Aug 10 15:42:23] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:42:23] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:42:24] DEBUG[17897] chan_sip.c: Allocating new SIP dialog for 7f90f4b664b0ba711f81d45b5e22eedd@127.0.0.1:5060 - OPTIONS (No RTP) [Aug 10 15:42:24] DEBUG[17897] acl.c: For destination '192.168.2.210', our source address is '192.168.1.84'. [Aug 10 15:42:24] DEBUG[17897] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.1.84:5060 [Aug 10 15:42:24] DEBUG[17897] chan_sip.c: Initializing initreq for method OPTIONS - callid 7738ec9a7b11eb2c4343fac94e7814df@192.168.1.84:5060 [Aug 10 15:42:24] DEBUG[17897] chan_sip.c: Header 0 [ 43]: OPTIONS sip:2210@192.168.2.210:2048 SIP/2.0 [Aug 10 15:42:24] DEBUG[17897] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK38fe09d4;rport [Aug 10 15:42:24] DEBUG[17897] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 10 15:42:24] DEBUG[17897] chan_sip.c: Header 3 [ 59]: From: "asterisk" ;tag=as09435ad3 [Aug 10 15:42:24] DEBUG[17897] chan_sip.c: Header 4 [ 33]: To: [Aug 10 15:42:24] DEBUG[17897] chan_sip.c: Header 5 [ 41]: Contact: [Aug 10 15:42:24] DEBUG[17897] chan_sip.c: Header 6 [ 59]: Call-ID: 7738ec9a7b11eb2c4343fac94e7814df@192.168.1.84:5060 [Aug 10 15:42:24] DEBUG[17897] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:42:24] DEBUG[17897] chan_sip.c: Header 8 [ 33]: User-Agent: Asterisk PBX 1.8.15.0 [Aug 10 15:42:24] DEBUG[17897] chan_sip.c: Header 9 [ 35]: Date: Fri, 10 Aug 2012 13:42:24 GMT [Aug 10 15:42:24] DEBUG[17897] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 10 15:42:24] DEBUG[17897] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 10 15:42:24] VERBOSE[17897] chan_sip.c: Reliably Transmitting (NAT) to 192.168.2.210:2048: OPTIONS sip:2210@192.168.2.210:2048 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK38fe09d4;rport Max-Forwards: 70 From: "asterisk" ;tag=as09435ad3 To: Contact: Call-ID: 7738ec9a7b11eb2c4343fac94e7814df@192.168.1.84:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.15.0 Date: Fri, 10 Aug 2012 13:42:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Aug 10 15:42:24] DEBUG[17897] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #235 [Aug 10 15:42:24] DEBUG[17897] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.2.210:2048 [Aug 10 15:42:24] VERBOSE[17897] chan_sip.c: <--- SIP read from UDP:192.168.2.210:2048 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK38fe09d4;rport=5060 From: "asterisk" ;tag=as09435ad3 To: Call-ID: 7738ec9a7b11eb2c4343fac94e7814df@192.168.1.84:5060 CSeq: 102 OPTIONS Contact: ;reg-id=1 User-Agent: snom300/8.4.32 Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: 100rel, replaces, from-change Content-Length: 0 <-------------> [Aug 10 15:42:24] DEBUG[17897] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 10 15:42:24] DEBUG[17897] chan_sip.c: Header 1 [ 68]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK38fe09d4;rport=5060 [Aug 10 15:42:24] DEBUG[17897] chan_sip.c: Header 2 [ 59]: From: "asterisk" ;tag=as09435ad3 [Aug 10 15:42:24] DEBUG[17897] chan_sip.c: Header 3 [ 33]: To: [Aug 10 15:42:24] DEBUG[17897] chan_sip.c: Header 4 [ 59]: Call-ID: 7738ec9a7b11eb2c4343fac94e7814df@192.168.1.84:5060 [Aug 10 15:42:24] DEBUG[17897] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:42:24] DEBUG[17897] chan_sip.c: Header 6 [ 47]: Contact: ;reg-id=1 [Aug 10 15:42:24] DEBUG[17897] chan_sip.c: Header 7 [ 26]: User-Agent: snom300/8.4.32 [Aug 10 15:42:24] DEBUG[17897] chan_sip.c: Header 8 [ 19]: Accept-Language: en [Aug 10 15:42:24] DEBUG[17897] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [Aug 10 15:42:24] DEBUG[17897] chan_sip.c: Header 10 [ 96]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE [Aug 10 15:42:24] DEBUG[17897] chan_sip.c: Header 11 [ 42]: Allow-Events: talk, hold, refer, call-info [Aug 10 15:42:24] DEBUG[17897] chan_sip.c: Header 12 [ 40]: Supported: 100rel, replaces, from-change [Aug 10 15:42:24] DEBUG[17897] chan_sip.c: Header 13 [ 17]: Content-Length: 0 [Aug 10 15:42:24] VERBOSE[17897] chan_sip.c: --- (14 headers 0 lines) --- [Aug 10 15:42:24] DEBUG[17897] chan_sip.c: = Looking for Call ID: 7738ec9a7b11eb2c4343fac94e7814df@192.168.1.84:5060 (Checking To) --From tag as09435ad3 --To-tag [Aug 10 15:42:24] DEBUG[17897] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #235 [Aug 10 15:42:24] DEBUG[17897] chan_sip.c: Stopping retransmission on '7738ec9a7b11eb2c4343fac94e7814df@192.168.1.84:5060' of Request 102: Match Found [Aug 10 15:42:24] DEBUG[17897] chan_sip.c: Destroying SIP dialog 7738ec9a7b11eb2c4343fac94e7814df@192.168.1.84:5060 [Aug 10 15:42:24] VERBOSE[17897] chan_sip.c: Really destroying SIP dialog '7738ec9a7b11eb2c4343fac94e7814df@192.168.1.84:5060' Method: OPTIONS [Aug 10 15:42:24] DEBUG[17897] chan_sip.c: Allocating new SIP dialog for 53c629d930d399a76842333504f5c64b@127.0.0.1:5060 - OPTIONS (No RTP) [Aug 10 15:42:24] DEBUG[17897] acl.c: For destination '192.168.1.102', our source address is '192.168.1.84'. [Aug 10 15:42:24] DEBUG[17897] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.1.84:5060 [Aug 10 15:42:24] DEBUG[17897] chan_sip.c: Initializing initreq for method OPTIONS - callid 2a367c0f7923b2d87ebf73603f129aa0@192.168.1.84:5060 [Aug 10 15:42:24] DEBUG[17897] chan_sip.c: Header 0 [ 57]: OPTIONS sip:2212@192.168.1.102:2048;line=z1zv7jsj SIP/2.0 [Aug 10 15:42:24] DEBUG[17897] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK37781f81;rport [Aug 10 15:42:24] DEBUG[17897] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 10 15:42:24] DEBUG[17897] chan_sip.c: Header 3 [ 59]: From: "asterisk" ;tag=as31e93ee9 [Aug 10 15:42:24] DEBUG[17897] chan_sip.c: Header 4 [ 47]: To: [Aug 10 15:42:24] DEBUG[17897] chan_sip.c: Header 5 [ 41]: Contact: [Aug 10 15:42:24] DEBUG[17897] chan_sip.c: Header 6 [ 59]: Call-ID: 2a367c0f7923b2d87ebf73603f129aa0@192.168.1.84:5060 [Aug 10 15:42:24] DEBUG[17897] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:42:24] DEBUG[17897] chan_sip.c: Header 8 [ 33]: User-Agent: Asterisk PBX 1.8.15.0 [Aug 10 15:42:24] DEBUG[17897] chan_sip.c: Header 9 [ 35]: Date: Fri, 10 Aug 2012 13:42:24 GMT [Aug 10 15:42:24] DEBUG[17897] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 10 15:42:24] DEBUG[17897] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 10 15:42:24] VERBOSE[17897] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.102:2048: OPTIONS sip:2212@192.168.1.102:2048;line=z1zv7jsj SIP/2.0 Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK37781f81;rport Max-Forwards: 70 From: "asterisk" ;tag=as31e93ee9 To: Contact: Call-ID: 2a367c0f7923b2d87ebf73603f129aa0@192.168.1.84:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.15.0 Date: Fri, 10 Aug 2012 13:42:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Aug 10 15:42:24] DEBUG[17897] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #238 [Aug 10 15:42:24] DEBUG[17897] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.1.102:2048 [Aug 10 15:42:24] VERBOSE[17897] chan_sip.c: <--- SIP read from UDP:192.168.1.102:2048 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK37781f81;rport=5060 From: "asterisk" ;tag=as31e93ee9 To: Call-ID: 2a367c0f7923b2d87ebf73603f129aa0@192.168.1.84:5060 CSeq: 102 OPTIONS Contact: ;reg-id=1 User-Agent: snom300/8.4.32 Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Content-Length: 0 <-------------> [Aug 10 15:42:24] DEBUG[17897] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 10 15:42:24] DEBUG[17897] chan_sip.c: Header 1 [ 68]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK37781f81;rport=5060 [Aug 10 15:42:24] DEBUG[17897] chan_sip.c: Header 2 [ 59]: From: "asterisk" ;tag=as31e93ee9 [Aug 10 15:42:24] DEBUG[17897] chan_sip.c: Header 3 [ 47]: To: [Aug 10 15:42:24] DEBUG[17897] chan_sip.c: Header 4 [ 59]: Call-ID: 2a367c0f7923b2d87ebf73603f129aa0@192.168.1.84:5060 [Aug 10 15:42:24] DEBUG[17897] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:42:24] DEBUG[17897] chan_sip.c: Header 6 [ 61]: Contact: ;reg-id=1 [Aug 10 15:42:24] DEBUG[17897] chan_sip.c: Header 7 [ 26]: User-Agent: snom300/8.4.32 [Aug 10 15:42:24] DEBUG[17897] chan_sip.c: Header 8 [ 19]: Accept-Language: en [Aug 10 15:42:24] DEBUG[17897] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [Aug 10 15:42:24] DEBUG[17897] chan_sip.c: Header 10 [ 96]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE [Aug 10 15:42:24] DEBUG[17897] chan_sip.c: Header 11 [ 42]: Allow-Events: talk, hold, refer, call-info [Aug 10 15:42:24] DEBUG[17897] chan_sip.c: Header 12 [ 47]: Supported: timer, 100rel, replaces, from-change [Aug 10 15:42:24] DEBUG[17897] chan_sip.c: Header 13 [ 17]: Content-Length: 0 [Aug 10 15:42:24] VERBOSE[17897] chan_sip.c: --- (14 headers 0 lines) --- [Aug 10 15:42:24] DEBUG[17897] chan_sip.c: = Looking for Call ID: 2a367c0f7923b2d87ebf73603f129aa0@192.168.1.84:5060 (Checking To) --From tag as31e93ee9 --To-tag [Aug 10 15:42:24] DEBUG[17897] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #238 [Aug 10 15:42:24] DEBUG[17897] chan_sip.c: Stopping retransmission on '2a367c0f7923b2d87ebf73603f129aa0@192.168.1.84:5060' of Request 102: Match Found [Aug 10 15:42:24] DEBUG[17897] chan_sip.c: Destroying SIP dialog 2a367c0f7923b2d87ebf73603f129aa0@192.168.1.84:5060 [Aug 10 15:42:24] VERBOSE[17897] chan_sip.c: Really destroying SIP dialog '2a367c0f7923b2d87ebf73603f129aa0@192.168.1.84:5060' Method: OPTIONS [Aug 10 15:42:28] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:42:28] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:42:33] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:42:33] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:42:38] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:42:38] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:42:43] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:42:43] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:42:44] DEBUG[17897] chan_sip.c: Allocating new SIP dialog for 71848786647612a15cec2dc90f83f528@127.0.0.1:5060 - OPTIONS (No RTP) [Aug 10 15:42:44] DEBUG[17897] acl.c: For destination '192.168.1.106', our source address is '192.168.1.84'. [Aug 10 15:42:44] DEBUG[17897] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.1.84:5060 [Aug 10 15:42:44] DEBUG[17897] chan_sip.c: Initializing initreq for method OPTIONS - callid 077d305a4ffe272a6f4f72593aafa185@192.168.1.84:5060 [Aug 10 15:42:44] DEBUG[17897] chan_sip.c: Header 0 [ 57]: OPTIONS sip:2219@192.168.1.106:2048;line=7g8ggsvj SIP/2.0 [Aug 10 15:42:44] DEBUG[17897] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK5c462a41;rport [Aug 10 15:42:44] DEBUG[17897] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 10 15:42:44] DEBUG[17897] chan_sip.c: Header 3 [ 59]: From: "asterisk" ;tag=as21968c4d [Aug 10 15:42:44] DEBUG[17897] chan_sip.c: Header 4 [ 47]: To: [Aug 10 15:42:44] DEBUG[17897] chan_sip.c: Header 5 [ 41]: Contact: [Aug 10 15:42:44] DEBUG[17897] chan_sip.c: Header 6 [ 59]: Call-ID: 077d305a4ffe272a6f4f72593aafa185@192.168.1.84:5060 [Aug 10 15:42:44] DEBUG[17897] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:42:44] DEBUG[17897] chan_sip.c: Header 8 [ 33]: User-Agent: Asterisk PBX 1.8.15.0 [Aug 10 15:42:44] DEBUG[17897] chan_sip.c: Header 9 [ 35]: Date: Fri, 10 Aug 2012 13:42:44 GMT [Aug 10 15:42:44] DEBUG[17897] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 10 15:42:44] DEBUG[17897] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 10 15:42:44] VERBOSE[17897] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.106:2048: OPTIONS sip:2219@192.168.1.106:2048;line=7g8ggsvj SIP/2.0 Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK5c462a41;rport Max-Forwards: 70 From: "asterisk" ;tag=as21968c4d To: Contact: Call-ID: 077d305a4ffe272a6f4f72593aafa185@192.168.1.84:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.15.0 Date: Fri, 10 Aug 2012 13:42:44 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Aug 10 15:42:44] DEBUG[17897] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #241 [Aug 10 15:42:44] DEBUG[17897] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.1.106:2048 [Aug 10 15:42:44] VERBOSE[17897] chan_sip.c: <--- SIP read from UDP:192.168.1.106:2048 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK5c462a41;rport=5060 From: "asterisk" ;tag=as21968c4d To: ;tag=7rknrgczxx Call-ID: 077d305a4ffe272a6f4f72593aafa185@192.168.1.84:5060 CSeq: 102 OPTIONS Contact: ;reg-id=1 User-Agent: snom300/8.7.3.10 Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: 100rel, replaces, from-change Content-Length: 0 <-------------> [Aug 10 15:42:44] DEBUG[17897] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 10 15:42:44] DEBUG[17897] chan_sip.c: Header 1 [ 68]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK5c462a41;rport=5060 [Aug 10 15:42:44] DEBUG[17897] chan_sip.c: Header 2 [ 59]: From: "asterisk" ;tag=as21968c4d [Aug 10 15:42:44] DEBUG[17897] chan_sip.c: Header 3 [ 62]: To: ;tag=7rknrgczxx [Aug 10 15:42:44] DEBUG[17897] chan_sip.c: Header 4 [ 59]: Call-ID: 077d305a4ffe272a6f4f72593aafa185@192.168.1.84:5060 [Aug 10 15:42:44] DEBUG[17897] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:42:44] DEBUG[17897] chan_sip.c: Header 6 [ 61]: Contact: ;reg-id=1 [Aug 10 15:42:44] DEBUG[17897] chan_sip.c: Header 7 [ 28]: User-Agent: snom300/8.7.3.10 [Aug 10 15:42:44] DEBUG[17897] chan_sip.c: Header 8 [ 19]: Accept-Language: en [Aug 10 15:42:44] DEBUG[17897] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [Aug 10 15:42:44] DEBUG[17897] chan_sip.c: Header 10 [ 96]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE [Aug 10 15:42:44] DEBUG[17897] chan_sip.c: Header 11 [ 42]: Allow-Events: talk, hold, refer, call-info [Aug 10 15:42:44] DEBUG[17897] chan_sip.c: Header 12 [ 40]: Supported: 100rel, replaces, from-change [Aug 10 15:42:44] DEBUG[17897] chan_sip.c: Header 13 [ 17]: Content-Length: 0 [Aug 10 15:42:44] VERBOSE[17897] chan_sip.c: --- (14 headers 0 lines) --- [Aug 10 15:42:44] DEBUG[17897] chan_sip.c: = Looking for Call ID: 077d305a4ffe272a6f4f72593aafa185@192.168.1.84:5060 (Checking To) --From tag as21968c4d --To-tag 7rknrgczxx [Aug 10 15:42:44] DEBUG[17897] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #241 [Aug 10 15:42:44] DEBUG[17897] chan_sip.c: Stopping retransmission on '077d305a4ffe272a6f4f72593aafa185@192.168.1.84:5060' of Request 102: Match Found [Aug 10 15:42:44] DEBUG[17897] chan_sip.c: Destroying SIP dialog 077d305a4ffe272a6f4f72593aafa185@192.168.1.84:5060 [Aug 10 15:42:44] VERBOSE[17897] chan_sip.c: Really destroying SIP dialog '077d305a4ffe272a6f4f72593aafa185@192.168.1.84:5060' Method: OPTIONS [Aug 10 15:42:45] VERBOSE[17897] chan_sip.c: <--- SIP read from UDP:192.168.1.106:2048 ---> REGISTER sip:192.168.1.84 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.106:2048;branch=z9hG4bK-xbi9wtswqq6q;rport From: "2219" ;tag=l55mkkxcsq To: "2219" Call-ID: 50250e9b9684-8d3g9c7o4hy2 CSeq: 17 REGISTER Max-Forwards: 70 Contact: ;reg-id=1;q=1.0;+sip.instance="";audio;mobility="fixed";duplex="full";description="snom300";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO" User-Agent: snom300/8.7.3.10 Allow-Events: dialog X-Real-IP: 192.168.1.106 Supported: path, gruu Content-Length: 0 <-------------> [Aug 10 15:42:45] DEBUG[17897] chan_sip.c: Header 0 [ 33]: REGISTER sip:192.168.1.84 SIP/2.0 [Aug 10 15:42:45] DEBUG[17897] chan_sip.c: Header 1 [ 69]: Via: SIP/2.0/UDP 192.168.1.106:2048;branch=z9hG4bK-xbi9wtswqq6q;rport [Aug 10 15:42:45] DEBUG[17897] chan_sip.c: Header 2 [ 51]: From: "2219" ;tag=l55mkkxcsq [Aug 10 15:42:45] DEBUG[17897] chan_sip.c: Header 3 [ 34]: To: "2219" [Aug 10 15:42:45] DEBUG[17897] chan_sip.c: Header 4 [ 34]: Call-ID: 50250e9b9684-8d3g9c7o4hy2 [Aug 10 15:42:45] DEBUG[17897] chan_sip.c: Header 5 [ 17]: CSeq: 17 REGISTER [Aug 10 15:42:45] DEBUG[17897] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Aug 10 15:42:45] DEBUG[17897] chan_sip.c: Header 7 [306]: Contact: ;reg-id=1;q=1.0;+sip.instance="";audio;mobility="fixed";duplex="full";description="snom300";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO" [Aug 10 15:42:45] DEBUG[17897] chan_sip.c: Header 8 [ 28]: User-Agent: snom300/8.7.3.10 [Aug 10 15:42:45] DEBUG[17897] chan_sip.c: Header 9 [ 20]: Allow-Events: dialog [Aug 10 15:42:45] DEBUG[17897] chan_sip.c: Header 10 [ 24]: X-Real-IP: 192.168.1.106 [Aug 10 15:42:45] DEBUG[17897] chan_sip.c: Header 11 [ 21]: Supported: path, gruu [Aug 10 15:42:45] DEBUG[17897] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Aug 10 15:42:45] VERBOSE[17897] chan_sip.c: --- (13 headers 0 lines) --- [Aug 10 15:42:45] DEBUG[17897] chan_sip.c: = Looking for Call ID: 50250e9b9684-8d3g9c7o4hy2 (Checking From) --From tag l55mkkxcsq --To-tag [Aug 10 15:42:45] DEBUG[17897] acl.c: For destination '192.168.1.106', our source address is '192.168.1.84'. [Aug 10 15:42:45] DEBUG[17897] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.1.84:5060 [Aug 10 15:42:45] DEBUG[17897] chan_sip.c: Allocating new SIP dialog for 50250e9b9684-8d3g9c7o4hy2 - REGISTER (No RTP) [Aug 10 15:42:45] DEBUG[17897] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Aug 10 15:42:45] DEBUG[17897] chan_sip.c: Initializing initreq for method REGISTER - callid 50250e9b9684-8d3g9c7o4hy2 [Aug 10 15:42:45] DEBUG[17897] netsock2.c: Splitting '192.168.1.106:2048' into... [Aug 10 15:42:45] DEBUG[17897] netsock2.c: ...host '192.168.1.106' and port '2048'. [Aug 10 15:42:45] VERBOSE[17897] chan_sip.c: Sending to 192.168.1.106:2048 (NAT) [Aug 10 15:42:45] DEBUG[17897] netsock2.c: Splitting '192.168.1.84' into... [Aug 10 15:42:45] DEBUG[17897] netsock2.c: ...host '192.168.1.84' and port ''. [Aug 10 15:42:45] VERBOSE[17897] chan_sip.c: <--- Transmitting (NAT) to 192.168.1.106:2048 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.106:2048;branch=z9hG4bK-xbi9wtswqq6q;received=192.168.1.106;rport=2048 From: "2219" ;tag=l55mkkxcsq To: "2219" ;tag=as3b7a4d18 Call-ID: 50250e9b9684-8d3g9c7o4hy2 CSeq: 17 REGISTER Server: Asterisk PBX 1.8.15.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="02e5b923" Content-Length: 0 <------------> [Aug 10 15:42:45] DEBUG[17897] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 192.168.1.106:2048 [Aug 10 15:42:45] VERBOSE[17897] chan_sip.c: Scheduling destruction of SIP dialog '50250e9b9684-8d3g9c7o4hy2' in 32000 ms (Method: REGISTER) [Aug 10 15:42:45] VERBOSE[17897] chan_sip.c: <--- SIP read from UDP:192.168.1.106:2048 ---> REGISTER sip:192.168.1.84 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.106:2048;branch=z9hG4bK-j49ju7ec0aps;rport From: "2219" ;tag=l55mkkxcsq To: "2219" Call-ID: 50250e9b9684-8d3g9c7o4hy2 CSeq: 18 REGISTER Max-Forwards: 70 Contact: ;reg-id=1;q=1.0;+sip.instance="";audio;mobility="fixed";duplex="full";description="snom300";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO" User-Agent: snom300/8.7.3.10 Allow-Events: dialog X-Real-IP: 192.168.1.106 Supported: path, gruu Authorization: Digest username="2219",realm="asterisk",nonce="02e5b923",uri="sip:192.168.1.84",response="c4f0c595152eb8e2d0bdb32ca94e97c0",algorithm=MD5 Content-Length: 0 <-------------> [Aug 10 15:42:45] DEBUG[17897] chan_sip.c: Header 0 [ 33]: REGISTER sip:192.168.1.84 SIP/2.0 [Aug 10 15:42:45] DEBUG[17897] chan_sip.c: Header 1 [ 69]: Via: SIP/2.0/UDP 192.168.1.106:2048;branch=z9hG4bK-j49ju7ec0aps;rport [Aug 10 15:42:45] DEBUG[17897] chan_sip.c: Header 2 [ 51]: From: "2219" ;tag=l55mkkxcsq [Aug 10 15:42:45] DEBUG[17897] chan_sip.c: Header 3 [ 34]: To: "2219" [Aug 10 15:42:45] DEBUG[17897] chan_sip.c: Header 4 [ 34]: Call-ID: 50250e9b9684-8d3g9c7o4hy2 [Aug 10 15:42:45] DEBUG[17897] chan_sip.c: Header 5 [ 17]: CSeq: 18 REGISTER [Aug 10 15:42:45] DEBUG[17897] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Aug 10 15:42:45] DEBUG[17897] chan_sip.c: Header 7 [306]: Contact: ;reg-id=1;q=1.0;+sip.instance="";audio;mobility="fixed";duplex="full";description="snom300";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO" [Aug 10 15:42:45] DEBUG[17897] chan_sip.c: Header 8 [ 28]: User-Agent: snom300/8.7.3.10 [Aug 10 15:42:45] DEBUG[17897] chan_sip.c: Header 9 [ 20]: Allow-Events: dialog [Aug 10 15:42:45] DEBUG[17897] chan_sip.c: Header 10 [ 24]: X-Real-IP: 192.168.1.106 [Aug 10 15:42:45] DEBUG[17897] chan_sip.c: Header 11 [ 21]: Supported: path, gruu [Aug 10 15:42:45] DEBUG[17897] chan_sip.c: Header 12 [152]: Authorization: Digest username="2219",realm="asterisk",nonce="02e5b923",uri="sip:192.168.1.84",response="c4f0c595152eb8e2d0bdb32ca94e97c0",algorithm=MD5 [Aug 10 15:42:45] DEBUG[17897] chan_sip.c: Header 13 [ 17]: Content-Length: 0 [Aug 10 15:42:45] VERBOSE[17897] chan_sip.c: --- (14 headers 0 lines) --- [Aug 10 15:42:45] DEBUG[17897] chan_sip.c: = Looking for Call ID: 50250e9b9684-8d3g9c7o4hy2 (Checking From) --From tag l55mkkxcsq --To-tag [Aug 10 15:42:45] DEBUG[17897] netsock2.c: Splitting '192.168.1.84' into... [Aug 10 15:42:45] DEBUG[17897] netsock2.c: ...host '192.168.1.84' and port ''. [Aug 10 15:42:45] DEBUG[17897] netsock2.c: Splitting '192.168.1.84' into... [Aug 10 15:42:45] DEBUG[17897] netsock2.c: ...host '192.168.1.84' and port ''. [Aug 10 15:42:45] DEBUG[17897] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Aug 10 15:42:45] DEBUG[17897] chan_sip.c: Initializing initreq for method REGISTER - callid 50250e9b9684-8d3g9c7o4hy2 [Aug 10 15:42:45] DEBUG[17897] netsock2.c: Splitting '192.168.1.106:2048' into... [Aug 10 15:42:45] DEBUG[17897] netsock2.c: ...host '192.168.1.106' and port '2048'. [Aug 10 15:42:45] VERBOSE[17897] chan_sip.c: Sending to 192.168.1.106:2048 (NAT) [Aug 10 15:42:45] DEBUG[17897] netsock2.c: Splitting '192.168.1.84' into... [Aug 10 15:42:45] DEBUG[17897] netsock2.c: ...host '192.168.1.84' and port ''. [Aug 10 15:42:45] DEBUG[17897] chan_sip.c: Store REGISTER's src-IP:port for call routing. [Aug 10 15:42:45] DEBUG[17897] chan_sip.c: Allocating new SIP dialog for 0e35584867ce9fc7676dc1506288f86e@127.0.0.1:5060 - OPTIONS (No RTP) [Aug 10 15:42:45] DEBUG[17897] acl.c: For destination '192.168.1.106', our source address is '192.168.1.84'. [Aug 10 15:42:45] DEBUG[17897] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.1.84:5060 [Aug 10 15:42:45] DEBUG[17897] chan_sip.c: Initializing initreq for method OPTIONS - callid 5faa05ef648d0b193eccea5c1327ca5b@192.168.1.84:5060 [Aug 10 15:42:45] DEBUG[17897] chan_sip.c: Header 0 [ 57]: OPTIONS sip:2219@192.168.1.106:2048;line=7g8ggsvj SIP/2.0 [Aug 10 15:42:45] DEBUG[17897] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK023010f6;rport [Aug 10 15:42:45] DEBUG[17897] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 10 15:42:45] DEBUG[17897] chan_sip.c: Header 3 [ 59]: From: "asterisk" ;tag=as5331766d [Aug 10 15:42:45] DEBUG[17897] chan_sip.c: Header 4 [ 47]: To: [Aug 10 15:42:45] DEBUG[17897] chan_sip.c: Header 5 [ 41]: Contact: [Aug 10 15:42:45] DEBUG[17897] chan_sip.c: Header 6 [ 59]: Call-ID: 5faa05ef648d0b193eccea5c1327ca5b@192.168.1.84:5060 [Aug 10 15:42:45] DEBUG[17897] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:42:45] DEBUG[17897] chan_sip.c: Header 8 [ 33]: User-Agent: Asterisk PBX 1.8.15.0 [Aug 10 15:42:45] DEBUG[17897] chan_sip.c: Header 9 [ 35]: Date: Fri, 10 Aug 2012 13:42:45 GMT [Aug 10 15:42:45] DEBUG[17897] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 10 15:42:45] DEBUG[17897] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 10 15:42:45] VERBOSE[17897] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.106:2048: OPTIONS sip:2219@192.168.1.106:2048;line=7g8ggsvj SIP/2.0 Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK023010f6;rport Max-Forwards: 70 From: "asterisk" ;tag=as5331766d To: Contact: Call-ID: 5faa05ef648d0b193eccea5c1327ca5b@192.168.1.84:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.15.0 Date: Fri, 10 Aug 2012 13:42:45 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Aug 10 15:42:45] DEBUG[17897] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #246 [Aug 10 15:42:45] DEBUG[17897] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.1.106:2048 [Aug 10 15:42:45] VERBOSE[17897] chan_sip.c: <--- Transmitting (NAT) to 192.168.1.106:2048 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.106:2048;branch=z9hG4bK-j49ju7ec0aps;received=192.168.1.106;rport=2048 From: "2219" ;tag=l55mkkxcsq To: "2219" ;tag=as3b7a4d18 Call-ID: 50250e9b9684-8d3g9c7o4hy2 CSeq: 18 REGISTER Server: Asterisk PBX 1.8.15.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Expires: 120 Contact: ;expires=120 Date: Fri, 10 Aug 2012 13:42:45 GMT Content-Length: 0 <------------> [Aug 10 15:42:45] DEBUG[17897] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.1.106:2048 [Aug 10 15:42:45] VERBOSE[17897] chan_sip.c: Scheduling destruction of SIP dialog '50250e9b9684-8d3g9c7o4hy2' in 32000 ms (Method: REGISTER) [Aug 10 15:42:45] DEBUG[17874] devicestate.c: No provider found, checking channel drivers for SIP - 2219 [Aug 10 15:42:45] DEBUG[17874] chan_sip.c: Checking device state for peer 2219 [Aug 10 15:42:45] DEBUG[17874] devicestate.c: Changing state for SIP/2219 - state 1 (Not in use) [Aug 10 15:42:45] DEBUG[17874] devicestate.c: device 'SIP/2219' state '1' [Aug 10 15:42:45] DEBUG[17907] app_queue.c: Device 'SIP/2219' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Aug 10 15:42:45] VERBOSE[17897] chan_sip.c: <--- SIP read from UDP:192.168.1.106:2048 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK023010f6;rport=5060 From: "asterisk" ;tag=as5331766d To: ;tag=2xyp2zcbsv Call-ID: 5faa05ef648d0b193eccea5c1327ca5b@192.168.1.84:5060 CSeq: 102 OPTIONS Contact: ;reg-id=1 User-Agent: snom300/8.7.3.10 Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: 100rel, replaces, from-change Content-Length: 0 <-------------> [Aug 10 15:42:45] DEBUG[17897] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 10 15:42:45] DEBUG[17897] chan_sip.c: Header 1 [ 68]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK023010f6;rport=5060 [Aug 10 15:42:45] DEBUG[17897] chan_sip.c: Header 2 [ 59]: From: "asterisk" ;tag=as5331766d [Aug 10 15:42:45] DEBUG[17897] chan_sip.c: Header 3 [ 62]: To: ;tag=2xyp2zcbsv [Aug 10 15:42:45] DEBUG[17897] chan_sip.c: Header 4 [ 59]: Call-ID: 5faa05ef648d0b193eccea5c1327ca5b@192.168.1.84:5060 [Aug 10 15:42:45] DEBUG[17897] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:42:45] DEBUG[17897] chan_sip.c: Header 6 [ 61]: Contact: ;reg-id=1 [Aug 10 15:42:45] DEBUG[17897] chan_sip.c: Header 7 [ 28]: User-Agent: snom300/8.7.3.10 [Aug 10 15:42:45] DEBUG[17897] chan_sip.c: Header 8 [ 19]: Accept-Language: en [Aug 10 15:42:45] DEBUG[17897] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [Aug 10 15:42:45] DEBUG[17897] chan_sip.c: Header 10 [ 96]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE [Aug 10 15:42:45] DEBUG[17897] chan_sip.c: Header 11 [ 42]: Allow-Events: talk, hold, refer, call-info [Aug 10 15:42:45] DEBUG[17897] chan_sip.c: Header 12 [ 40]: Supported: 100rel, replaces, from-change [Aug 10 15:42:45] DEBUG[17897] chan_sip.c: Header 13 [ 17]: Content-Length: 0 [Aug 10 15:42:45] VERBOSE[17897] chan_sip.c: --- (14 headers 0 lines) --- [Aug 10 15:42:45] DEBUG[17897] chan_sip.c: = Looking for Call ID: 5faa05ef648d0b193eccea5c1327ca5b@192.168.1.84:5060 (Checking To) --From tag as5331766d --To-tag 2xyp2zcbsv [Aug 10 15:42:45] DEBUG[17897] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #246 [Aug 10 15:42:45] DEBUG[17897] chan_sip.c: Stopping retransmission on '5faa05ef648d0b193eccea5c1327ca5b@192.168.1.84:5060' of Request 102: Match Found [Aug 10 15:42:45] DEBUG[17897] chan_sip.c: Destroying SIP dialog 5faa05ef648d0b193eccea5c1327ca5b@192.168.1.84:5060 [Aug 10 15:42:45] VERBOSE[17897] chan_sip.c: Really destroying SIP dialog '5faa05ef648d0b193eccea5c1327ca5b@192.168.1.84:5060' Method: OPTIONS [Aug 10 15:42:48] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:42:48] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:42:53] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:42:53] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:42:58] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:42:58] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:43:03] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:43:03] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:43:08] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:43:08] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:43:13] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:43:14] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:43:17] DEBUG[17897] chan_sip.c: Auto destroying SIP dialog '50250e9b9684-8d3g9c7o4hy2' [Aug 10 15:43:17] DEBUG[17897] chan_sip.c: Destroying SIP dialog 50250e9b9684-8d3g9c7o4hy2 [Aug 10 15:43:17] VERBOSE[17897] chan_sip.c: Really destroying SIP dialog '50250e9b9684-8d3g9c7o4hy2' Method: REGISTER [Aug 10 15:43:18] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:43:19] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:43:23] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:43:24] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:43:24] DEBUG[17897] chan_sip.c: Allocating new SIP dialog for 3a85f297496fdffb683f91bb39f24011@127.0.0.1:5060 - OPTIONS (No RTP) [Aug 10 15:43:24] DEBUG[17897] acl.c: For destination '192.168.2.210', our source address is '192.168.1.84'. [Aug 10 15:43:24] DEBUG[17897] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.1.84:5060 [Aug 10 15:43:24] DEBUG[17897] chan_sip.c: Initializing initreq for method OPTIONS - callid 0ccf0a896466fb79410678e8605832d2@192.168.1.84:5060 [Aug 10 15:43:24] DEBUG[17897] chan_sip.c: Header 0 [ 43]: OPTIONS sip:2210@192.168.2.210:2048 SIP/2.0 [Aug 10 15:43:24] DEBUG[17897] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK3d7f63c3;rport [Aug 10 15:43:24] DEBUG[17897] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 10 15:43:24] DEBUG[17897] chan_sip.c: Header 3 [ 59]: From: "asterisk" ;tag=as0bb7c6c9 [Aug 10 15:43:24] DEBUG[17897] chan_sip.c: Header 4 [ 33]: To: [Aug 10 15:43:24] DEBUG[17897] chan_sip.c: Header 5 [ 41]: Contact: [Aug 10 15:43:24] DEBUG[17897] chan_sip.c: Header 6 [ 59]: Call-ID: 0ccf0a896466fb79410678e8605832d2@192.168.1.84:5060 [Aug 10 15:43:24] DEBUG[17897] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:43:24] DEBUG[17897] chan_sip.c: Header 8 [ 33]: User-Agent: Asterisk PBX 1.8.15.0 [Aug 10 15:43:24] DEBUG[17897] chan_sip.c: Header 9 [ 35]: Date: Fri, 10 Aug 2012 13:43:24 GMT [Aug 10 15:43:24] DEBUG[17897] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 10 15:43:24] DEBUG[17897] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 10 15:43:24] VERBOSE[17897] chan_sip.c: Reliably Transmitting (NAT) to 192.168.2.210:2048: OPTIONS sip:2210@192.168.2.210:2048 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK3d7f63c3;rport Max-Forwards: 70 From: "asterisk" ;tag=as0bb7c6c9 To: Contact: Call-ID: 0ccf0a896466fb79410678e8605832d2@192.168.1.84:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.15.0 Date: Fri, 10 Aug 2012 13:43:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Aug 10 15:43:24] DEBUG[17897] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #250 [Aug 10 15:43:24] DEBUG[17897] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.2.210:2048 [Aug 10 15:43:24] VERBOSE[17897] chan_sip.c: <--- SIP read from UDP:192.168.2.210:2048 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK3d7f63c3;rport=5060 From: "asterisk" ;tag=as0bb7c6c9 To: Call-ID: 0ccf0a896466fb79410678e8605832d2@192.168.1.84:5060 CSeq: 102 OPTIONS Contact: ;reg-id=1 User-Agent: snom300/8.4.32 Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: 100rel, replaces, from-change Content-Length: 0 <-------------> [Aug 10 15:43:24] DEBUG[17897] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 10 15:43:24] DEBUG[17897] chan_sip.c: Header 1 [ 68]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK3d7f63c3;rport=5060 [Aug 10 15:43:24] DEBUG[17897] chan_sip.c: Header 2 [ 59]: From: "asterisk" ;tag=as0bb7c6c9 [Aug 10 15:43:24] DEBUG[17897] chan_sip.c: Header 3 [ 33]: To: [Aug 10 15:43:24] DEBUG[17897] chan_sip.c: Header 4 [ 59]: Call-ID: 0ccf0a896466fb79410678e8605832d2@192.168.1.84:5060 [Aug 10 15:43:24] DEBUG[17897] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:43:24] DEBUG[17897] chan_sip.c: Header 6 [ 47]: Contact: ;reg-id=1 [Aug 10 15:43:24] DEBUG[17897] chan_sip.c: Header 7 [ 26]: User-Agent: snom300/8.4.32 [Aug 10 15:43:24] DEBUG[17897] chan_sip.c: Header 8 [ 19]: Accept-Language: en [Aug 10 15:43:24] DEBUG[17897] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [Aug 10 15:43:24] DEBUG[17897] chan_sip.c: Header 10 [ 96]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE [Aug 10 15:43:24] DEBUG[17897] chan_sip.c: Header 11 [ 42]: Allow-Events: talk, hold, refer, call-info [Aug 10 15:43:24] DEBUG[17897] chan_sip.c: Header 12 [ 40]: Supported: 100rel, replaces, from-change [Aug 10 15:43:24] DEBUG[17897] chan_sip.c: Header 13 [ 17]: Content-Length: 0 [Aug 10 15:43:24] VERBOSE[17897] chan_sip.c: --- (14 headers 0 lines) --- [Aug 10 15:43:24] DEBUG[17897] chan_sip.c: = Looking for Call ID: 0ccf0a896466fb79410678e8605832d2@192.168.1.84:5060 (Checking To) --From tag as0bb7c6c9 --To-tag [Aug 10 15:43:24] DEBUG[17897] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #250 [Aug 10 15:43:24] DEBUG[17897] chan_sip.c: Stopping retransmission on '0ccf0a896466fb79410678e8605832d2@192.168.1.84:5060' of Request 102: Match Found [Aug 10 15:43:24] DEBUG[17897] chan_sip.c: Destroying SIP dialog 0ccf0a896466fb79410678e8605832d2@192.168.1.84:5060 [Aug 10 15:43:24] VERBOSE[17897] chan_sip.c: Really destroying SIP dialog '0ccf0a896466fb79410678e8605832d2@192.168.1.84:5060' Method: OPTIONS [Aug 10 15:43:24] DEBUG[17897] chan_sip.c: Allocating new SIP dialog for 1260b1622dc6708813aa54170fcedbf2@127.0.0.1:5060 - OPTIONS (No RTP) [Aug 10 15:43:24] DEBUG[17897] acl.c: For destination '192.168.1.102', our source address is '192.168.1.84'. [Aug 10 15:43:24] DEBUG[17897] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.1.84:5060 [Aug 10 15:43:24] DEBUG[17897] chan_sip.c: Initializing initreq for method OPTIONS - callid 081ef1fb5f7c6bf83ab04a2d2becad3d@192.168.1.84:5060 [Aug 10 15:43:24] DEBUG[17897] chan_sip.c: Header 0 [ 57]: OPTIONS sip:2212@192.168.1.102:2048;line=z1zv7jsj SIP/2.0 [Aug 10 15:43:24] DEBUG[17897] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK2b7bf3aa;rport [Aug 10 15:43:24] DEBUG[17897] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 10 15:43:24] DEBUG[17897] chan_sip.c: Header 3 [ 59]: From: "asterisk" ;tag=as5a621957 [Aug 10 15:43:24] DEBUG[17897] chan_sip.c: Header 4 [ 47]: To: [Aug 10 15:43:24] DEBUG[17897] chan_sip.c: Header 5 [ 41]: Contact: [Aug 10 15:43:24] DEBUG[17897] chan_sip.c: Header 6 [ 59]: Call-ID: 081ef1fb5f7c6bf83ab04a2d2becad3d@192.168.1.84:5060 [Aug 10 15:43:24] DEBUG[17897] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:43:24] DEBUG[17897] chan_sip.c: Header 8 [ 33]: User-Agent: Asterisk PBX 1.8.15.0 [Aug 10 15:43:24] DEBUG[17897] chan_sip.c: Header 9 [ 35]: Date: Fri, 10 Aug 2012 13:43:24 GMT [Aug 10 15:43:24] DEBUG[17897] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 10 15:43:24] DEBUG[17897] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 10 15:43:24] VERBOSE[17897] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.102:2048: OPTIONS sip:2212@192.168.1.102:2048;line=z1zv7jsj SIP/2.0 Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK2b7bf3aa;rport Max-Forwards: 70 From: "asterisk" ;tag=as5a621957 To: Contact: Call-ID: 081ef1fb5f7c6bf83ab04a2d2becad3d@192.168.1.84:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.15.0 Date: Fri, 10 Aug 2012 13:43:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Aug 10 15:43:24] DEBUG[17897] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #253 [Aug 10 15:43:24] DEBUG[17897] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.1.102:2048 [Aug 10 15:43:24] VERBOSE[17897] chan_sip.c: <--- SIP read from UDP:192.168.1.102:2048 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK2b7bf3aa;rport=5060 From: "asterisk" ;tag=as5a621957 To: Call-ID: 081ef1fb5f7c6bf83ab04a2d2becad3d@192.168.1.84:5060 CSeq: 102 OPTIONS Contact: ;reg-id=1 User-Agent: snom300/8.4.32 Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Content-Length: 0 <-------------> [Aug 10 15:43:24] DEBUG[17897] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 10 15:43:24] DEBUG[17897] chan_sip.c: Header 1 [ 68]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK2b7bf3aa;rport=5060 [Aug 10 15:43:24] DEBUG[17897] chan_sip.c: Header 2 [ 59]: From: "asterisk" ;tag=as5a621957 [Aug 10 15:43:24] DEBUG[17897] chan_sip.c: Header 3 [ 47]: To: [Aug 10 15:43:24] DEBUG[17897] chan_sip.c: Header 4 [ 59]: Call-ID: 081ef1fb5f7c6bf83ab04a2d2becad3d@192.168.1.84:5060 [Aug 10 15:43:24] DEBUG[17897] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:43:24] DEBUG[17897] chan_sip.c: Header 6 [ 61]: Contact: ;reg-id=1 [Aug 10 15:43:24] DEBUG[17897] chan_sip.c: Header 7 [ 26]: User-Agent: snom300/8.4.32 [Aug 10 15:43:24] DEBUG[17897] chan_sip.c: Header 8 [ 19]: Accept-Language: en [Aug 10 15:43:24] DEBUG[17897] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [Aug 10 15:43:24] DEBUG[17897] chan_sip.c: Header 10 [ 96]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE [Aug 10 15:43:24] DEBUG[17897] chan_sip.c: Header 11 [ 42]: Allow-Events: talk, hold, refer, call-info [Aug 10 15:43:24] DEBUG[17897] chan_sip.c: Header 12 [ 47]: Supported: timer, 100rel, replaces, from-change [Aug 10 15:43:24] DEBUG[17897] chan_sip.c: Header 13 [ 17]: Content-Length: 0 [Aug 10 15:43:24] VERBOSE[17897] chan_sip.c: --- (14 headers 0 lines) --- [Aug 10 15:43:24] DEBUG[17897] chan_sip.c: = Looking for Call ID: 081ef1fb5f7c6bf83ab04a2d2becad3d@192.168.1.84:5060 (Checking To) --From tag as5a621957 --To-tag [Aug 10 15:43:24] DEBUG[17897] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #253 [Aug 10 15:43:24] DEBUG[17897] chan_sip.c: Stopping retransmission on '081ef1fb5f7c6bf83ab04a2d2becad3d@192.168.1.84:5060' of Request 102: Match Found [Aug 10 15:43:24] DEBUG[17897] chan_sip.c: Destroying SIP dialog 081ef1fb5f7c6bf83ab04a2d2becad3d@192.168.1.84:5060 [Aug 10 15:43:24] VERBOSE[17897] chan_sip.c: Really destroying SIP dialog '081ef1fb5f7c6bf83ab04a2d2becad3d@192.168.1.84:5060' Method: OPTIONS [Aug 10 15:43:28] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:43:29] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:43:33] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:43:34] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:43:38] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:43:39] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:43:43] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:43:44] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: Allocating new SIP dialog for 1b48a2724252c37770f1a47143a21647@127.0.0.1:5060 - OPTIONS (No RTP) [Aug 10 15:43:45] DEBUG[17897] acl.c: For destination '192.168.1.106', our source address is '192.168.1.84'. [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.1.84:5060 [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: Initializing initreq for method OPTIONS - callid 12b7d777671368a638fb528f7da1f36e@192.168.1.84:5060 [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: Header 0 [ 57]: OPTIONS sip:2219@192.168.1.106:2048;line=7g8ggsvj SIP/2.0 [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK52e0a3fb;rport [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: Header 3 [ 59]: From: "asterisk" ;tag=as2f1b4d51 [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: Header 4 [ 47]: To: [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: Header 5 [ 41]: Contact: [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: Header 6 [ 59]: Call-ID: 12b7d777671368a638fb528f7da1f36e@192.168.1.84:5060 [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: Header 8 [ 33]: User-Agent: Asterisk PBX 1.8.15.0 [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: Header 9 [ 35]: Date: Fri, 10 Aug 2012 13:43:45 GMT [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 10 15:43:45] VERBOSE[17897] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.106:2048: OPTIONS sip:2219@192.168.1.106:2048;line=7g8ggsvj SIP/2.0 Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK52e0a3fb;rport Max-Forwards: 70 From: "asterisk" ;tag=as2f1b4d51 To: Contact: Call-ID: 12b7d777671368a638fb528f7da1f36e@192.168.1.84:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.15.0 Date: Fri, 10 Aug 2012 13:43:45 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #256 [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.1.106:2048 [Aug 10 15:43:45] VERBOSE[17897] chan_sip.c: <--- SIP read from UDP:192.168.1.106:2048 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK52e0a3fb;rport=5060 From: "asterisk" ;tag=as2f1b4d51 To: ;tag=gyte9bqla8 Call-ID: 12b7d777671368a638fb528f7da1f36e@192.168.1.84:5060 CSeq: 102 OPTIONS Contact: ;reg-id=1 User-Agent: snom300/8.7.3.10 Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: 100rel, replaces, from-change Content-Length: 0 <-------------> [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: Header 1 [ 68]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK52e0a3fb;rport=5060 [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: Header 2 [ 59]: From: "asterisk" ;tag=as2f1b4d51 [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: Header 3 [ 62]: To: ;tag=gyte9bqla8 [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: Header 4 [ 59]: Call-ID: 12b7d777671368a638fb528f7da1f36e@192.168.1.84:5060 [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: Header 6 [ 61]: Contact: ;reg-id=1 [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: Header 7 [ 28]: User-Agent: snom300/8.7.3.10 [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: Header 8 [ 19]: Accept-Language: en [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: Header 10 [ 96]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: Header 11 [ 42]: Allow-Events: talk, hold, refer, call-info [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: Header 12 [ 40]: Supported: 100rel, replaces, from-change [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: Header 13 [ 17]: Content-Length: 0 [Aug 10 15:43:45] VERBOSE[17897] chan_sip.c: --- (14 headers 0 lines) --- [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: = Looking for Call ID: 12b7d777671368a638fb528f7da1f36e@192.168.1.84:5060 (Checking To) --From tag as2f1b4d51 --To-tag gyte9bqla8 [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #256 [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: Stopping retransmission on '12b7d777671368a638fb528f7da1f36e@192.168.1.84:5060' of Request 102: Match Found [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: Destroying SIP dialog 12b7d777671368a638fb528f7da1f36e@192.168.1.84:5060 [Aug 10 15:43:45] VERBOSE[17897] chan_sip.c: Really destroying SIP dialog '12b7d777671368a638fb528f7da1f36e@192.168.1.84:5060' Method: OPTIONS [Aug 10 15:43:45] VERBOSE[17897] chan_sip.c: <--- SIP read from UDP:192.168.1.106:2048 ---> REGISTER sip:192.168.1.84 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.106:2048;branch=z9hG4bK-laoc51sz55xe;rport From: "2219" ;tag=6c44hm4tdc To: "2219" Call-ID: 50250e9b9684-8d3g9c7o4hy2 CSeq: 19 REGISTER Max-Forwards: 70 Contact: ;reg-id=1;q=1.0;+sip.instance="";audio;mobility="fixed";duplex="full";description="snom300";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO" User-Agent: snom300/8.7.3.10 Allow-Events: dialog X-Real-IP: 192.168.1.106 Supported: path, gruu Content-Length: 0 <-------------> [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: Header 0 [ 33]: REGISTER sip:192.168.1.84 SIP/2.0 [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: Header 1 [ 69]: Via: SIP/2.0/UDP 192.168.1.106:2048;branch=z9hG4bK-laoc51sz55xe;rport [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: Header 2 [ 51]: From: "2219" ;tag=6c44hm4tdc [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: Header 3 [ 34]: To: "2219" [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: Header 4 [ 34]: Call-ID: 50250e9b9684-8d3g9c7o4hy2 [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: Header 5 [ 17]: CSeq: 19 REGISTER [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: Header 7 [306]: Contact: ;reg-id=1;q=1.0;+sip.instance="";audio;mobility="fixed";duplex="full";description="snom300";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO" [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: Header 8 [ 28]: User-Agent: snom300/8.7.3.10 [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: Header 9 [ 20]: Allow-Events: dialog [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: Header 10 [ 24]: X-Real-IP: 192.168.1.106 [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: Header 11 [ 21]: Supported: path, gruu [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Aug 10 15:43:45] VERBOSE[17897] chan_sip.c: --- (13 headers 0 lines) --- [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: = Looking for Call ID: 50250e9b9684-8d3g9c7o4hy2 (Checking From) --From tag 6c44hm4tdc --To-tag [Aug 10 15:43:45] DEBUG[17897] acl.c: For destination '192.168.1.106', our source address is '192.168.1.84'. [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.1.84:5060 [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: Allocating new SIP dialog for 50250e9b9684-8d3g9c7o4hy2 - REGISTER (No RTP) [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: Initializing initreq for method REGISTER - callid 50250e9b9684-8d3g9c7o4hy2 [Aug 10 15:43:45] DEBUG[17897] netsock2.c: Splitting '192.168.1.106:2048' into... [Aug 10 15:43:45] DEBUG[17897] netsock2.c: ...host '192.168.1.106' and port '2048'. [Aug 10 15:43:45] VERBOSE[17897] chan_sip.c: Sending to 192.168.1.106:2048 (NAT) [Aug 10 15:43:45] DEBUG[17897] netsock2.c: Splitting '192.168.1.84' into... [Aug 10 15:43:45] DEBUG[17897] netsock2.c: ...host '192.168.1.84' and port ''. [Aug 10 15:43:45] VERBOSE[17897] chan_sip.c: <--- Transmitting (NAT) to 192.168.1.106:2048 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.106:2048;branch=z9hG4bK-laoc51sz55xe;received=192.168.1.106;rport=2048 From: "2219" ;tag=6c44hm4tdc To: "2219" ;tag=as46445f93 Call-ID: 50250e9b9684-8d3g9c7o4hy2 CSeq: 19 REGISTER Server: Asterisk PBX 1.8.15.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="63990a91" Content-Length: 0 <------------> [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 192.168.1.106:2048 [Aug 10 15:43:45] VERBOSE[17897] chan_sip.c: Scheduling destruction of SIP dialog '50250e9b9684-8d3g9c7o4hy2' in 32000 ms (Method: REGISTER) [Aug 10 15:43:45] VERBOSE[17897] chan_sip.c: <--- SIP read from UDP:192.168.1.106:2048 ---> REGISTER sip:192.168.1.84 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.106:2048;branch=z9hG4bK-x9bqpwlx8qik;rport From: "2219" ;tag=6c44hm4tdc To: "2219" Call-ID: 50250e9b9684-8d3g9c7o4hy2 CSeq: 20 REGISTER Max-Forwards: 70 Contact: ;reg-id=1;q=1.0;+sip.instance="";audio;mobility="fixed";duplex="full";description="snom300";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO" User-Agent: snom300/8.7.3.10 Allow-Events: dialog X-Real-IP: 192.168.1.106 Supported: path, gruu Authorization: Digest username="2219",realm="asterisk",nonce="63990a91",uri="sip:192.168.1.84",response="5f99801f19711364a93a776cb5fe9828",algorithm=MD5 Content-Length: 0 <-------------> [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: Header 0 [ 33]: REGISTER sip:192.168.1.84 SIP/2.0 [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: Header 1 [ 69]: Via: SIP/2.0/UDP 192.168.1.106:2048;branch=z9hG4bK-x9bqpwlx8qik;rport [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: Header 2 [ 51]: From: "2219" ;tag=6c44hm4tdc [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: Header 3 [ 34]: To: "2219" [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: Header 4 [ 34]: Call-ID: 50250e9b9684-8d3g9c7o4hy2 [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: Header 5 [ 17]: CSeq: 20 REGISTER [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: Header 7 [306]: Contact: ;reg-id=1;q=1.0;+sip.instance="";audio;mobility="fixed";duplex="full";description="snom300";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO" [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: Header 8 [ 28]: User-Agent: snom300/8.7.3.10 [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: Header 9 [ 20]: Allow-Events: dialog [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: Header 10 [ 24]: X-Real-IP: 192.168.1.106 [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: Header 11 [ 21]: Supported: path, gruu [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: Header 12 [152]: Authorization: Digest username="2219",realm="asterisk",nonce="63990a91",uri="sip:192.168.1.84",response="5f99801f19711364a93a776cb5fe9828",algorithm=MD5 [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: Header 13 [ 17]: Content-Length: 0 [Aug 10 15:43:45] VERBOSE[17897] chan_sip.c: --- (14 headers 0 lines) --- [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: = Looking for Call ID: 50250e9b9684-8d3g9c7o4hy2 (Checking From) --From tag 6c44hm4tdc --To-tag [Aug 10 15:43:45] DEBUG[17897] netsock2.c: Splitting '192.168.1.84' into... [Aug 10 15:43:45] DEBUG[17897] netsock2.c: ...host '192.168.1.84' and port ''. [Aug 10 15:43:45] DEBUG[17897] netsock2.c: Splitting '192.168.1.84' into... [Aug 10 15:43:45] DEBUG[17897] netsock2.c: ...host '192.168.1.84' and port ''. [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: Initializing initreq for method REGISTER - callid 50250e9b9684-8d3g9c7o4hy2 [Aug 10 15:43:45] DEBUG[17897] netsock2.c: Splitting '192.168.1.106:2048' into... [Aug 10 15:43:45] DEBUG[17897] netsock2.c: ...host '192.168.1.106' and port '2048'. [Aug 10 15:43:45] VERBOSE[17897] chan_sip.c: Sending to 192.168.1.106:2048 (NAT) [Aug 10 15:43:45] DEBUG[17897] netsock2.c: Splitting '192.168.1.84' into... [Aug 10 15:43:45] DEBUG[17897] netsock2.c: ...host '192.168.1.84' and port ''. [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: Store REGISTER's src-IP:port for call routing. [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: Allocating new SIP dialog for 1aae254d133b89500f1d89ed41170561@127.0.0.1:5060 - OPTIONS (No RTP) [Aug 10 15:43:45] DEBUG[17897] acl.c: For destination '192.168.1.106', our source address is '192.168.1.84'. [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.1.84:5060 [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: Initializing initreq for method OPTIONS - callid 2f2012c82bbd17f84a84b42b52ea397d@192.168.1.84:5060 [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: Header 0 [ 57]: OPTIONS sip:2219@192.168.1.106:2048;line=7g8ggsvj SIP/2.0 [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK6ca9ef8f;rport [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: Header 3 [ 59]: From: "asterisk" ;tag=as5ac1795d [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: Header 4 [ 47]: To: [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: Header 5 [ 41]: Contact: [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: Header 6 [ 59]: Call-ID: 2f2012c82bbd17f84a84b42b52ea397d@192.168.1.84:5060 [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: Header 8 [ 33]: User-Agent: Asterisk PBX 1.8.15.0 [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: Header 9 [ 35]: Date: Fri, 10 Aug 2012 13:43:45 GMT [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 10 15:43:45] VERBOSE[17897] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.106:2048: OPTIONS sip:2219@192.168.1.106:2048;line=7g8ggsvj SIP/2.0 Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK6ca9ef8f;rport Max-Forwards: 70 From: "asterisk" ;tag=as5ac1795d To: Contact: Call-ID: 2f2012c82bbd17f84a84b42b52ea397d@192.168.1.84:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.15.0 Date: Fri, 10 Aug 2012 13:43:45 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #261 [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.1.106:2048 [Aug 10 15:43:45] VERBOSE[17897] chan_sip.c: <--- Transmitting (NAT) to 192.168.1.106:2048 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.106:2048;branch=z9hG4bK-x9bqpwlx8qik;received=192.168.1.106;rport=2048 From: "2219" ;tag=6c44hm4tdc To: "2219" ;tag=as46445f93 Call-ID: 50250e9b9684-8d3g9c7o4hy2 CSeq: 20 REGISTER Server: Asterisk PBX 1.8.15.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Expires: 120 Contact: ;expires=120 Date: Fri, 10 Aug 2012 13:43:45 GMT Content-Length: 0 <------------> [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.1.106:2048 [Aug 10 15:43:45] VERBOSE[17897] chan_sip.c: Scheduling destruction of SIP dialog '50250e9b9684-8d3g9c7o4hy2' in 32000 ms (Method: REGISTER) [Aug 10 15:43:45] DEBUG[17874] devicestate.c: No provider found, checking channel drivers for SIP - 2219 [Aug 10 15:43:45] DEBUG[17874] chan_sip.c: Checking device state for peer 2219 [Aug 10 15:43:45] DEBUG[17874] devicestate.c: Changing state for SIP/2219 - state 1 (Not in use) [Aug 10 15:43:45] DEBUG[17874] devicestate.c: device 'SIP/2219' state '1' [Aug 10 15:43:45] DEBUG[17907] app_queue.c: Device 'SIP/2219' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Aug 10 15:43:45] VERBOSE[17897] chan_sip.c: <--- SIP read from UDP:192.168.1.106:2048 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK6ca9ef8f;rport=5060 From: "asterisk" ;tag=as5ac1795d To: ;tag=q3pv5x9mty Call-ID: 2f2012c82bbd17f84a84b42b52ea397d@192.168.1.84:5060 CSeq: 102 OPTIONS Contact: ;reg-id=1 User-Agent: snom300/8.7.3.10 Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: 100rel, replaces, from-change Content-Length: 0 <-------------> [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: Header 1 [ 68]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK6ca9ef8f;rport=5060 [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: Header 2 [ 59]: From: "asterisk" ;tag=as5ac1795d [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: Header 3 [ 62]: To: ;tag=q3pv5x9mty [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: Header 4 [ 59]: Call-ID: 2f2012c82bbd17f84a84b42b52ea397d@192.168.1.84:5060 [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: Header 6 [ 61]: Contact: ;reg-id=1 [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: Header 7 [ 28]: User-Agent: snom300/8.7.3.10 [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: Header 8 [ 19]: Accept-Language: en [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: Header 10 [ 96]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: Header 11 [ 42]: Allow-Events: talk, hold, refer, call-info [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: Header 12 [ 40]: Supported: 100rel, replaces, from-change [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: Header 13 [ 17]: Content-Length: 0 [Aug 10 15:43:45] VERBOSE[17897] chan_sip.c: --- (14 headers 0 lines) --- [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: = Looking for Call ID: 2f2012c82bbd17f84a84b42b52ea397d@192.168.1.84:5060 (Checking To) --From tag as5ac1795d --To-tag q3pv5x9mty [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #261 [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: Stopping retransmission on '2f2012c82bbd17f84a84b42b52ea397d@192.168.1.84:5060' of Request 102: Match Found [Aug 10 15:43:45] DEBUG[17897] chan_sip.c: Destroying SIP dialog 2f2012c82bbd17f84a84b42b52ea397d@192.168.1.84:5060 [Aug 10 15:43:45] VERBOSE[17897] chan_sip.c: Really destroying SIP dialog '2f2012c82bbd17f84a84b42b52ea397d@192.168.1.84:5060' Method: OPTIONS [Aug 10 15:43:48] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:43:49] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:43:53] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:43:54] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:43:58] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:43:59] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:44:03] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:44:04] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:44:08] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:44:09] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:44:13] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:44:14] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:44:17] DEBUG[17897] chan_sip.c: Auto destroying SIP dialog '50250e9b9684-8d3g9c7o4hy2' [Aug 10 15:44:17] DEBUG[17897] chan_sip.c: Destroying SIP dialog 50250e9b9684-8d3g9c7o4hy2 [Aug 10 15:44:17] VERBOSE[17897] chan_sip.c: Really destroying SIP dialog '50250e9b9684-8d3g9c7o4hy2' Method: REGISTER [Aug 10 15:44:19] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:44:19] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:44:24] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:44:24] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:44:24] DEBUG[17897] chan_sip.c: Allocating new SIP dialog for 56bd63ce40ad72c40519cfd81ab585a0@127.0.0.1:5060 - OPTIONS (No RTP) [Aug 10 15:44:24] DEBUG[17897] acl.c: For destination '192.168.2.210', our source address is '192.168.1.84'. [Aug 10 15:44:24] DEBUG[17897] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.1.84:5060 [Aug 10 15:44:24] DEBUG[17897] chan_sip.c: Initializing initreq for method OPTIONS - callid 4c4f496633e44c05369fa69949577c5d@192.168.1.84:5060 [Aug 10 15:44:24] DEBUG[17897] chan_sip.c: Header 0 [ 43]: OPTIONS sip:2210@192.168.2.210:2048 SIP/2.0 [Aug 10 15:44:24] DEBUG[17897] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK5454ac21;rport [Aug 10 15:44:24] DEBUG[17897] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 10 15:44:24] DEBUG[17897] chan_sip.c: Header 3 [ 59]: From: "asterisk" ;tag=as14845f4c [Aug 10 15:44:24] DEBUG[17897] chan_sip.c: Header 4 [ 33]: To: [Aug 10 15:44:24] DEBUG[17897] chan_sip.c: Header 5 [ 41]: Contact: [Aug 10 15:44:24] DEBUG[17897] chan_sip.c: Header 6 [ 59]: Call-ID: 4c4f496633e44c05369fa69949577c5d@192.168.1.84:5060 [Aug 10 15:44:24] DEBUG[17897] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:44:24] DEBUG[17897] chan_sip.c: Header 8 [ 33]: User-Agent: Asterisk PBX 1.8.15.0 [Aug 10 15:44:24] DEBUG[17897] chan_sip.c: Header 9 [ 35]: Date: Fri, 10 Aug 2012 13:44:24 GMT [Aug 10 15:44:24] DEBUG[17897] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 10 15:44:24] DEBUG[17897] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 10 15:44:24] VERBOSE[17897] chan_sip.c: Reliably Transmitting (NAT) to 192.168.2.210:2048: OPTIONS sip:2210@192.168.2.210:2048 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK5454ac21;rport Max-Forwards: 70 From: "asterisk" ;tag=as14845f4c To: Contact: Call-ID: 4c4f496633e44c05369fa69949577c5d@192.168.1.84:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.15.0 Date: Fri, 10 Aug 2012 13:44:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Aug 10 15:44:24] DEBUG[17897] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #265 [Aug 10 15:44:24] DEBUG[17897] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.2.210:2048 [Aug 10 15:44:24] VERBOSE[17897] chan_sip.c: <--- SIP read from UDP:192.168.2.210:2048 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK5454ac21;rport=5060 From: "asterisk" ;tag=as14845f4c To: Call-ID: 4c4f496633e44c05369fa69949577c5d@192.168.1.84:5060 CSeq: 102 OPTIONS Contact: ;reg-id=1 User-Agent: snom300/8.4.32 Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: 100rel, replaces, from-change Content-Length: 0 <-------------> [Aug 10 15:44:24] DEBUG[17897] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 10 15:44:24] DEBUG[17897] chan_sip.c: Header 1 [ 68]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK5454ac21;rport=5060 [Aug 10 15:44:24] DEBUG[17897] chan_sip.c: Header 2 [ 59]: From: "asterisk" ;tag=as14845f4c [Aug 10 15:44:24] DEBUG[17897] chan_sip.c: Header 3 [ 33]: To: [Aug 10 15:44:24] DEBUG[17897] chan_sip.c: Header 4 [ 59]: Call-ID: 4c4f496633e44c05369fa69949577c5d@192.168.1.84:5060 [Aug 10 15:44:24] DEBUG[17897] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:44:24] DEBUG[17897] chan_sip.c: Header 6 [ 47]: Contact: ;reg-id=1 [Aug 10 15:44:24] DEBUG[17897] chan_sip.c: Header 7 [ 26]: User-Agent: snom300/8.4.32 [Aug 10 15:44:24] DEBUG[17897] chan_sip.c: Header 8 [ 19]: Accept-Language: en [Aug 10 15:44:24] DEBUG[17897] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [Aug 10 15:44:24] DEBUG[17897] chan_sip.c: Header 10 [ 96]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE [Aug 10 15:44:24] DEBUG[17897] chan_sip.c: Header 11 [ 42]: Allow-Events: talk, hold, refer, call-info [Aug 10 15:44:24] DEBUG[17897] chan_sip.c: Header 12 [ 40]: Supported: 100rel, replaces, from-change [Aug 10 15:44:24] DEBUG[17897] chan_sip.c: Header 13 [ 17]: Content-Length: 0 [Aug 10 15:44:24] VERBOSE[17897] chan_sip.c: --- (14 headers 0 lines) --- [Aug 10 15:44:24] DEBUG[17897] chan_sip.c: = Looking for Call ID: 4c4f496633e44c05369fa69949577c5d@192.168.1.84:5060 (Checking To) --From tag as14845f4c --To-tag [Aug 10 15:44:24] DEBUG[17897] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #265 [Aug 10 15:44:24] DEBUG[17897] chan_sip.c: Stopping retransmission on '4c4f496633e44c05369fa69949577c5d@192.168.1.84:5060' of Request 102: Match Found [Aug 10 15:44:24] DEBUG[17897] chan_sip.c: Destroying SIP dialog 4c4f496633e44c05369fa69949577c5d@192.168.1.84:5060 [Aug 10 15:44:24] VERBOSE[17897] chan_sip.c: Really destroying SIP dialog '4c4f496633e44c05369fa69949577c5d@192.168.1.84:5060' Method: OPTIONS [Aug 10 15:44:24] DEBUG[17897] chan_sip.c: Allocating new SIP dialog for 3271b21846bc7d5625196c95054d9aa7@127.0.0.1:5060 - OPTIONS (No RTP) [Aug 10 15:44:24] DEBUG[17897] acl.c: For destination '192.168.1.102', our source address is '192.168.1.84'. [Aug 10 15:44:24] DEBUG[17897] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.1.84:5060 [Aug 10 15:44:24] DEBUG[17897] chan_sip.c: Initializing initreq for method OPTIONS - callid 770c73b04d7db5473212a9500877185b@192.168.1.84:5060 [Aug 10 15:44:24] DEBUG[17897] chan_sip.c: Header 0 [ 57]: OPTIONS sip:2212@192.168.1.102:2048;line=z1zv7jsj SIP/2.0 [Aug 10 15:44:24] DEBUG[17897] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK50283d4b;rport [Aug 10 15:44:24] DEBUG[17897] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 10 15:44:24] DEBUG[17897] chan_sip.c: Header 3 [ 59]: From: "asterisk" ;tag=as10971c08 [Aug 10 15:44:24] DEBUG[17897] chan_sip.c: Header 4 [ 47]: To: [Aug 10 15:44:24] DEBUG[17897] chan_sip.c: Header 5 [ 41]: Contact: [Aug 10 15:44:24] DEBUG[17897] chan_sip.c: Header 6 [ 59]: Call-ID: 770c73b04d7db5473212a9500877185b@192.168.1.84:5060 [Aug 10 15:44:24] DEBUG[17897] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:44:24] DEBUG[17897] chan_sip.c: Header 8 [ 33]: User-Agent: Asterisk PBX 1.8.15.0 [Aug 10 15:44:24] DEBUG[17897] chan_sip.c: Header 9 [ 35]: Date: Fri, 10 Aug 2012 13:44:24 GMT [Aug 10 15:44:24] DEBUG[17897] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 10 15:44:24] DEBUG[17897] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 10 15:44:24] VERBOSE[17897] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.102:2048: OPTIONS sip:2212@192.168.1.102:2048;line=z1zv7jsj SIP/2.0 Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK50283d4b;rport Max-Forwards: 70 From: "asterisk" ;tag=as10971c08 To: Contact: Call-ID: 770c73b04d7db5473212a9500877185b@192.168.1.84:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.15.0 Date: Fri, 10 Aug 2012 13:44:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Aug 10 15:44:24] DEBUG[17897] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #268 [Aug 10 15:44:24] DEBUG[17897] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.1.102:2048 [Aug 10 15:44:24] VERBOSE[17897] chan_sip.c: <--- SIP read from UDP:192.168.1.102:2048 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK50283d4b;rport=5060 From: "asterisk" ;tag=as10971c08 To: Call-ID: 770c73b04d7db5473212a9500877185b@192.168.1.84:5060 CSeq: 102 OPTIONS Contact: ;reg-id=1 User-Agent: snom300/8.4.32 Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Content-Length: 0 <-------------> [Aug 10 15:44:24] DEBUG[17897] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 10 15:44:24] DEBUG[17897] chan_sip.c: Header 1 [ 68]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK50283d4b;rport=5060 [Aug 10 15:44:24] DEBUG[17897] chan_sip.c: Header 2 [ 59]: From: "asterisk" ;tag=as10971c08 [Aug 10 15:44:24] DEBUG[17897] chan_sip.c: Header 3 [ 47]: To: [Aug 10 15:44:24] DEBUG[17897] chan_sip.c: Header 4 [ 59]: Call-ID: 770c73b04d7db5473212a9500877185b@192.168.1.84:5060 [Aug 10 15:44:24] DEBUG[17897] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:44:24] DEBUG[17897] chan_sip.c: Header 6 [ 61]: Contact: ;reg-id=1 [Aug 10 15:44:24] DEBUG[17897] chan_sip.c: Header 7 [ 26]: User-Agent: snom300/8.4.32 [Aug 10 15:44:24] DEBUG[17897] chan_sip.c: Header 8 [ 19]: Accept-Language: en [Aug 10 15:44:24] DEBUG[17897] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [Aug 10 15:44:24] DEBUG[17897] chan_sip.c: Header 10 [ 96]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE [Aug 10 15:44:24] DEBUG[17897] chan_sip.c: Header 11 [ 42]: Allow-Events: talk, hold, refer, call-info [Aug 10 15:44:24] DEBUG[17897] chan_sip.c: Header 12 [ 47]: Supported: timer, 100rel, replaces, from-change [Aug 10 15:44:24] DEBUG[17897] chan_sip.c: Header 13 [ 17]: Content-Length: 0 [Aug 10 15:44:24] VERBOSE[17897] chan_sip.c: --- (14 headers 0 lines) --- [Aug 10 15:44:24] DEBUG[17897] chan_sip.c: = Looking for Call ID: 770c73b04d7db5473212a9500877185b@192.168.1.84:5060 (Checking To) --From tag as10971c08 --To-tag [Aug 10 15:44:24] DEBUG[17897] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #268 [Aug 10 15:44:24] DEBUG[17897] chan_sip.c: Stopping retransmission on '770c73b04d7db5473212a9500877185b@192.168.1.84:5060' of Request 102: Match Found [Aug 10 15:44:24] DEBUG[17897] chan_sip.c: Destroying SIP dialog 770c73b04d7db5473212a9500877185b@192.168.1.84:5060 [Aug 10 15:44:24] VERBOSE[17897] chan_sip.c: Really destroying SIP dialog '770c73b04d7db5473212a9500877185b@192.168.1.84:5060' Method: OPTIONS [Aug 10 15:44:29] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:44:29] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:44:34] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:44:34] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:44:39] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:44:39] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:44:44] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:44:44] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: Allocating new SIP dialog for 4deb6faa5696f545362a773630453144@127.0.0.1:5060 - OPTIONS (No RTP) [Aug 10 15:44:45] DEBUG[17897] acl.c: For destination '192.168.1.106', our source address is '192.168.1.84'. [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.1.84:5060 [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: Initializing initreq for method OPTIONS - callid 18198c4a32f13c425ff9d73a71cbaaa3@192.168.1.84:5060 [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: Header 0 [ 57]: OPTIONS sip:2219@192.168.1.106:2048;line=7g8ggsvj SIP/2.0 [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK56e24c51;rport [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: Header 3 [ 59]: From: "asterisk" ;tag=as0c186542 [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: Header 4 [ 47]: To: [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: Header 5 [ 41]: Contact: [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: Header 6 [ 59]: Call-ID: 18198c4a32f13c425ff9d73a71cbaaa3@192.168.1.84:5060 [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: Header 8 [ 33]: User-Agent: Asterisk PBX 1.8.15.0 [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: Header 9 [ 35]: Date: Fri, 10 Aug 2012 13:44:45 GMT [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 10 15:44:45] VERBOSE[17897] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.106:2048: OPTIONS sip:2219@192.168.1.106:2048;line=7g8ggsvj SIP/2.0 Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK56e24c51;rport Max-Forwards: 70 From: "asterisk" ;tag=as0c186542 To: Contact: Call-ID: 18198c4a32f13c425ff9d73a71cbaaa3@192.168.1.84:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.15.0 Date: Fri, 10 Aug 2012 13:44:45 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #271 [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.1.106:2048 [Aug 10 15:44:45] VERBOSE[17897] chan_sip.c: <--- SIP read from UDP:192.168.1.106:2048 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK56e24c51;rport=5060 From: "asterisk" ;tag=as0c186542 To: ;tag=eyazmbxqaa Call-ID: 18198c4a32f13c425ff9d73a71cbaaa3@192.168.1.84:5060 CSeq: 102 OPTIONS Contact: ;reg-id=1 User-Agent: snom300/8.7.3.10 Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: 100rel, replaces, from-change Content-Length: 0 <-------------> [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: Header 1 [ 68]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK56e24c51;rport=5060 [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: Header 2 [ 59]: From: "asterisk" ;tag=as0c186542 [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: Header 3 [ 62]: To: ;tag=eyazmbxqaa [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: Header 4 [ 59]: Call-ID: 18198c4a32f13c425ff9d73a71cbaaa3@192.168.1.84:5060 [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: Header 6 [ 61]: Contact: ;reg-id=1 [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: Header 7 [ 28]: User-Agent: snom300/8.7.3.10 [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: Header 8 [ 19]: Accept-Language: en [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: Header 10 [ 96]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: Header 11 [ 42]: Allow-Events: talk, hold, refer, call-info [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: Header 12 [ 40]: Supported: 100rel, replaces, from-change [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: Header 13 [ 17]: Content-Length: 0 [Aug 10 15:44:45] VERBOSE[17897] chan_sip.c: --- (14 headers 0 lines) --- [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: = Looking for Call ID: 18198c4a32f13c425ff9d73a71cbaaa3@192.168.1.84:5060 (Checking To) --From tag as0c186542 --To-tag eyazmbxqaa [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #271 [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: Stopping retransmission on '18198c4a32f13c425ff9d73a71cbaaa3@192.168.1.84:5060' of Request 102: Match Found [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: Destroying SIP dialog 18198c4a32f13c425ff9d73a71cbaaa3@192.168.1.84:5060 [Aug 10 15:44:45] VERBOSE[17897] chan_sip.c: Really destroying SIP dialog '18198c4a32f13c425ff9d73a71cbaaa3@192.168.1.84:5060' Method: OPTIONS [Aug 10 15:44:45] VERBOSE[17897] chan_sip.c: <--- SIP read from UDP:192.168.1.106:2048 ---> REGISTER sip:192.168.1.84 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.106:2048;branch=z9hG4bK-iidouee5fmdq;rport From: "2219" ;tag=m0ubbfok1w To: "2219" Call-ID: 50250e9b9684-8d3g9c7o4hy2 CSeq: 21 REGISTER Max-Forwards: 70 Contact: ;reg-id=1;q=1.0;+sip.instance="";audio;mobility="fixed";duplex="full";description="snom300";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO" User-Agent: snom300/8.7.3.10 Allow-Events: dialog X-Real-IP: 192.168.1.106 Supported: path, gruu Content-Length: 0 <-------------> [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: Header 0 [ 33]: REGISTER sip:192.168.1.84 SIP/2.0 [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: Header 1 [ 69]: Via: SIP/2.0/UDP 192.168.1.106:2048;branch=z9hG4bK-iidouee5fmdq;rport [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: Header 2 [ 51]: From: "2219" ;tag=m0ubbfok1w [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: Header 3 [ 34]: To: "2219" [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: Header 4 [ 34]: Call-ID: 50250e9b9684-8d3g9c7o4hy2 [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: Header 5 [ 17]: CSeq: 21 REGISTER [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: Header 7 [306]: Contact: ;reg-id=1;q=1.0;+sip.instance="";audio;mobility="fixed";duplex="full";description="snom300";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO" [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: Header 8 [ 28]: User-Agent: snom300/8.7.3.10 [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: Header 9 [ 20]: Allow-Events: dialog [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: Header 10 [ 24]: X-Real-IP: 192.168.1.106 [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: Header 11 [ 21]: Supported: path, gruu [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Aug 10 15:44:45] VERBOSE[17897] chan_sip.c: --- (13 headers 0 lines) --- [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: = Looking for Call ID: 50250e9b9684-8d3g9c7o4hy2 (Checking From) --From tag m0ubbfok1w --To-tag [Aug 10 15:44:45] DEBUG[17897] acl.c: For destination '192.168.1.106', our source address is '192.168.1.84'. [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.1.84:5060 [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: Allocating new SIP dialog for 50250e9b9684-8d3g9c7o4hy2 - REGISTER (No RTP) [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: Initializing initreq for method REGISTER - callid 50250e9b9684-8d3g9c7o4hy2 [Aug 10 15:44:45] DEBUG[17897] netsock2.c: Splitting '192.168.1.106:2048' into... [Aug 10 15:44:45] DEBUG[17897] netsock2.c: ...host '192.168.1.106' and port '2048'. [Aug 10 15:44:45] VERBOSE[17897] chan_sip.c: Sending to 192.168.1.106:2048 (NAT) [Aug 10 15:44:45] DEBUG[17897] netsock2.c: Splitting '192.168.1.84' into... [Aug 10 15:44:45] DEBUG[17897] netsock2.c: ...host '192.168.1.84' and port ''. [Aug 10 15:44:45] VERBOSE[17897] chan_sip.c: <--- Transmitting (NAT) to 192.168.1.106:2048 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.106:2048;branch=z9hG4bK-iidouee5fmdq;received=192.168.1.106;rport=2048 From: "2219" ;tag=m0ubbfok1w To: "2219" ;tag=as75e3f838 Call-ID: 50250e9b9684-8d3g9c7o4hy2 CSeq: 21 REGISTER Server: Asterisk PBX 1.8.15.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5f8054e1" Content-Length: 0 <------------> [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 192.168.1.106:2048 [Aug 10 15:44:45] VERBOSE[17897] chan_sip.c: Scheduling destruction of SIP dialog '50250e9b9684-8d3g9c7o4hy2' in 32000 ms (Method: REGISTER) [Aug 10 15:44:45] VERBOSE[17897] chan_sip.c: <--- SIP read from UDP:192.168.1.106:2048 ---> REGISTER sip:192.168.1.84 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.106:2048;branch=z9hG4bK-6v831rmef0cu;rport From: "2219" ;tag=m0ubbfok1w To: "2219" Call-ID: 50250e9b9684-8d3g9c7o4hy2 CSeq: 22 REGISTER Max-Forwards: 70 Contact: ;reg-id=1;q=1.0;+sip.instance="";audio;mobility="fixed";duplex="full";description="snom300";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO" User-Agent: snom300/8.7.3.10 Allow-Events: dialog X-Real-IP: 192.168.1.106 Supported: path, gruu Authorization: Digest username="2219",realm="asterisk",nonce="5f8054e1",uri="sip:192.168.1.84",response="0b04df8916a298cef0999cce443ba5ed",algorithm=MD5 Content-Length: 0 <-------------> [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: Header 0 [ 33]: REGISTER sip:192.168.1.84 SIP/2.0 [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: Header 1 [ 69]: Via: SIP/2.0/UDP 192.168.1.106:2048;branch=z9hG4bK-6v831rmef0cu;rport [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: Header 2 [ 51]: From: "2219" ;tag=m0ubbfok1w [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: Header 3 [ 34]: To: "2219" [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: Header 4 [ 34]: Call-ID: 50250e9b9684-8d3g9c7o4hy2 [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: Header 5 [ 17]: CSeq: 22 REGISTER [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: Header 7 [306]: Contact: ;reg-id=1;q=1.0;+sip.instance="";audio;mobility="fixed";duplex="full";description="snom300";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO" [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: Header 8 [ 28]: User-Agent: snom300/8.7.3.10 [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: Header 9 [ 20]: Allow-Events: dialog [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: Header 10 [ 24]: X-Real-IP: 192.168.1.106 [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: Header 11 [ 21]: Supported: path, gruu [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: Header 12 [152]: Authorization: Digest username="2219",realm="asterisk",nonce="5f8054e1",uri="sip:192.168.1.84",response="0b04df8916a298cef0999cce443ba5ed",algorithm=MD5 [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: Header 13 [ 17]: Content-Length: 0 [Aug 10 15:44:45] VERBOSE[17897] chan_sip.c: --- (14 headers 0 lines) --- [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: = Looking for Call ID: 50250e9b9684-8d3g9c7o4hy2 (Checking From) --From tag m0ubbfok1w --To-tag [Aug 10 15:44:45] DEBUG[17897] netsock2.c: Splitting '192.168.1.84' into... [Aug 10 15:44:45] DEBUG[17897] netsock2.c: ...host '192.168.1.84' and port ''. [Aug 10 15:44:45] DEBUG[17897] netsock2.c: Splitting '192.168.1.84' into... [Aug 10 15:44:45] DEBUG[17897] netsock2.c: ...host '192.168.1.84' and port ''. [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: Initializing initreq for method REGISTER - callid 50250e9b9684-8d3g9c7o4hy2 [Aug 10 15:44:45] DEBUG[17897] netsock2.c: Splitting '192.168.1.106:2048' into... [Aug 10 15:44:45] DEBUG[17897] netsock2.c: ...host '192.168.1.106' and port '2048'. [Aug 10 15:44:45] VERBOSE[17897] chan_sip.c: Sending to 192.168.1.106:2048 (NAT) [Aug 10 15:44:45] DEBUG[17897] netsock2.c: Splitting '192.168.1.84' into... [Aug 10 15:44:45] DEBUG[17897] netsock2.c: ...host '192.168.1.84' and port ''. [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: Store REGISTER's src-IP:port for call routing. [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: Allocating new SIP dialog for 47b8c44727a973472295b0357d250aa8@127.0.0.1:5060 - OPTIONS (No RTP) [Aug 10 15:44:45] DEBUG[17897] acl.c: For destination '192.168.1.106', our source address is '192.168.1.84'. [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.1.84:5060 [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: Initializing initreq for method OPTIONS - callid 4173e2ee2ddec035308ecb510952b77f@192.168.1.84:5060 [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: Header 0 [ 57]: OPTIONS sip:2219@192.168.1.106:2048;line=7g8ggsvj SIP/2.0 [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK54a410c4;rport [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: Header 3 [ 59]: From: "asterisk" ;tag=as7169a79b [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: Header 4 [ 47]: To: [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: Header 5 [ 41]: Contact: [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: Header 6 [ 59]: Call-ID: 4173e2ee2ddec035308ecb510952b77f@192.168.1.84:5060 [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: Header 8 [ 33]: User-Agent: Asterisk PBX 1.8.15.0 [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: Header 9 [ 35]: Date: Fri, 10 Aug 2012 13:44:45 GMT [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 10 15:44:45] VERBOSE[17897] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.106:2048: OPTIONS sip:2219@192.168.1.106:2048;line=7g8ggsvj SIP/2.0 Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK54a410c4;rport Max-Forwards: 70 From: "asterisk" ;tag=as7169a79b To: Contact: Call-ID: 4173e2ee2ddec035308ecb510952b77f@192.168.1.84:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.15.0 Date: Fri, 10 Aug 2012 13:44:45 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #276 [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.1.106:2048 [Aug 10 15:44:45] VERBOSE[17897] chan_sip.c: <--- Transmitting (NAT) to 192.168.1.106:2048 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.106:2048;branch=z9hG4bK-6v831rmef0cu;received=192.168.1.106;rport=2048 From: "2219" ;tag=m0ubbfok1w To: "2219" ;tag=as75e3f838 Call-ID: 50250e9b9684-8d3g9c7o4hy2 CSeq: 22 REGISTER Server: Asterisk PBX 1.8.15.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Expires: 120 Contact: ;expires=120 Date: Fri, 10 Aug 2012 13:44:45 GMT Content-Length: 0 <------------> [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.1.106:2048 [Aug 10 15:44:45] VERBOSE[17897] chan_sip.c: Scheduling destruction of SIP dialog '50250e9b9684-8d3g9c7o4hy2' in 32000 ms (Method: REGISTER) [Aug 10 15:44:45] DEBUG[17874] devicestate.c: No provider found, checking channel drivers for SIP - 2219 [Aug 10 15:44:45] DEBUG[17874] chan_sip.c: Checking device state for peer 2219 [Aug 10 15:44:45] DEBUG[17874] devicestate.c: Changing state for SIP/2219 - state 1 (Not in use) [Aug 10 15:44:45] DEBUG[17874] devicestate.c: device 'SIP/2219' state '1' [Aug 10 15:44:45] DEBUG[17907] app_queue.c: Device 'SIP/2219' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Aug 10 15:44:45] VERBOSE[17897] chan_sip.c: <--- SIP read from UDP:192.168.1.106:2048 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK54a410c4;rport=5060 From: "asterisk" ;tag=as7169a79b To: ;tag=2c0rqcr3r3 Call-ID: 4173e2ee2ddec035308ecb510952b77f@192.168.1.84:5060 CSeq: 102 OPTIONS Contact: ;reg-id=1 User-Agent: snom300/8.7.3.10 Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: 100rel, replaces, from-change Content-Length: 0 <-------------> [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: Header 1 [ 68]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK54a410c4;rport=5060 [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: Header 2 [ 59]: From: "asterisk" ;tag=as7169a79b [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: Header 3 [ 62]: To: ;tag=2c0rqcr3r3 [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: Header 4 [ 59]: Call-ID: 4173e2ee2ddec035308ecb510952b77f@192.168.1.84:5060 [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: Header 6 [ 61]: Contact: ;reg-id=1 [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: Header 7 [ 28]: User-Agent: snom300/8.7.3.10 [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: Header 8 [ 19]: Accept-Language: en [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: Header 10 [ 96]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: Header 11 [ 42]: Allow-Events: talk, hold, refer, call-info [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: Header 12 [ 40]: Supported: 100rel, replaces, from-change [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: Header 13 [ 17]: Content-Length: 0 [Aug 10 15:44:45] VERBOSE[17897] chan_sip.c: --- (14 headers 0 lines) --- [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: = Looking for Call ID: 4173e2ee2ddec035308ecb510952b77f@192.168.1.84:5060 (Checking To) --From tag as7169a79b --To-tag 2c0rqcr3r3 [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #276 [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: Stopping retransmission on '4173e2ee2ddec035308ecb510952b77f@192.168.1.84:5060' of Request 102: Match Found [Aug 10 15:44:45] DEBUG[17897] chan_sip.c: Destroying SIP dialog 4173e2ee2ddec035308ecb510952b77f@192.168.1.84:5060 [Aug 10 15:44:45] VERBOSE[17897] chan_sip.c: Really destroying SIP dialog '4173e2ee2ddec035308ecb510952b77f@192.168.1.84:5060' Method: OPTIONS [Aug 10 15:44:49] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:44:49] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:44:54] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:44:54] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:44:59] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:44:59] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:45:04] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:45:04] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:45:09] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:45:09] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:45:14] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:45:14] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:45:17] DEBUG[17897] chan_sip.c: Auto destroying SIP dialog '50250e9b9684-8d3g9c7o4hy2' [Aug 10 15:45:17] DEBUG[17897] chan_sip.c: Destroying SIP dialog 50250e9b9684-8d3g9c7o4hy2 [Aug 10 15:45:17] VERBOSE[17897] chan_sip.c: Really destroying SIP dialog '50250e9b9684-8d3g9c7o4hy2' Method: REGISTER [Aug 10 15:45:19] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:45:19] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:45:24] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:45:24] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:45:24] DEBUG[17897] chan_sip.c: Allocating new SIP dialog for 2f8645e96603a63b09e059c97440c7bf@127.0.0.1:5060 - OPTIONS (No RTP) [Aug 10 15:45:24] DEBUG[17897] acl.c: For destination '192.168.2.210', our source address is '192.168.1.84'. [Aug 10 15:45:24] DEBUG[17897] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.1.84:5060 [Aug 10 15:45:24] DEBUG[17897] chan_sip.c: Initializing initreq for method OPTIONS - callid 01a170f034f987986891b5fc7a6455b5@192.168.1.84:5060 [Aug 10 15:45:24] DEBUG[17897] chan_sip.c: Header 0 [ 43]: OPTIONS sip:2210@192.168.2.210:2048 SIP/2.0 [Aug 10 15:45:24] DEBUG[17897] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK409d584a;rport [Aug 10 15:45:24] DEBUG[17897] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 10 15:45:24] DEBUG[17897] chan_sip.c: Header 3 [ 59]: From: "asterisk" ;tag=as667de343 [Aug 10 15:45:24] DEBUG[17897] chan_sip.c: Header 4 [ 33]: To: [Aug 10 15:45:24] DEBUG[17897] chan_sip.c: Header 5 [ 41]: Contact: [Aug 10 15:45:24] DEBUG[17897] chan_sip.c: Header 6 [ 59]: Call-ID: 01a170f034f987986891b5fc7a6455b5@192.168.1.84:5060 [Aug 10 15:45:24] DEBUG[17897] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:45:24] DEBUG[17897] chan_sip.c: Header 8 [ 33]: User-Agent: Asterisk PBX 1.8.15.0 [Aug 10 15:45:24] DEBUG[17897] chan_sip.c: Header 9 [ 35]: Date: Fri, 10 Aug 2012 13:45:24 GMT [Aug 10 15:45:24] DEBUG[17897] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 10 15:45:24] DEBUG[17897] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 10 15:45:24] VERBOSE[17897] chan_sip.c: Reliably Transmitting (NAT) to 192.168.2.210:2048: OPTIONS sip:2210@192.168.2.210:2048 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK409d584a;rport Max-Forwards: 70 From: "asterisk" ;tag=as667de343 To: Contact: Call-ID: 01a170f034f987986891b5fc7a6455b5@192.168.1.84:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.15.0 Date: Fri, 10 Aug 2012 13:45:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Aug 10 15:45:24] DEBUG[17897] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #280 [Aug 10 15:45:24] DEBUG[17897] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.2.210:2048 [Aug 10 15:45:24] VERBOSE[17897] chan_sip.c: <--- SIP read from UDP:192.168.2.210:2048 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK409d584a;rport=5060 From: "asterisk" ;tag=as667de343 To: Call-ID: 01a170f034f987986891b5fc7a6455b5@192.168.1.84:5060 CSeq: 102 OPTIONS Contact: ;reg-id=1 User-Agent: snom300/8.4.32 Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: 100rel, replaces, from-change Content-Length: 0 <-------------> [Aug 10 15:45:24] DEBUG[17897] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 10 15:45:24] DEBUG[17897] chan_sip.c: Header 1 [ 68]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK409d584a;rport=5060 [Aug 10 15:45:24] DEBUG[17897] chan_sip.c: Header 2 [ 59]: From: "asterisk" ;tag=as667de343 [Aug 10 15:45:24] DEBUG[17897] chan_sip.c: Header 3 [ 33]: To: [Aug 10 15:45:24] DEBUG[17897] chan_sip.c: Header 4 [ 59]: Call-ID: 01a170f034f987986891b5fc7a6455b5@192.168.1.84:5060 [Aug 10 15:45:24] DEBUG[17897] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:45:24] DEBUG[17897] chan_sip.c: Header 6 [ 47]: Contact: ;reg-id=1 [Aug 10 15:45:24] DEBUG[17897] chan_sip.c: Header 7 [ 26]: User-Agent: snom300/8.4.32 [Aug 10 15:45:24] DEBUG[17897] chan_sip.c: Header 8 [ 19]: Accept-Language: en [Aug 10 15:45:24] DEBUG[17897] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [Aug 10 15:45:24] DEBUG[17897] chan_sip.c: Header 10 [ 96]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE [Aug 10 15:45:24] DEBUG[17897] chan_sip.c: Header 11 [ 42]: Allow-Events: talk, hold, refer, call-info [Aug 10 15:45:24] DEBUG[17897] chan_sip.c: Header 12 [ 40]: Supported: 100rel, replaces, from-change [Aug 10 15:45:24] DEBUG[17897] chan_sip.c: Header 13 [ 17]: Content-Length: 0 [Aug 10 15:45:24] VERBOSE[17897] chan_sip.c: --- (14 headers 0 lines) --- [Aug 10 15:45:24] DEBUG[17897] chan_sip.c: = Looking for Call ID: 01a170f034f987986891b5fc7a6455b5@192.168.1.84:5060 (Checking To) --From tag as667de343 --To-tag [Aug 10 15:45:24] DEBUG[17897] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #280 [Aug 10 15:45:24] DEBUG[17897] chan_sip.c: Stopping retransmission on '01a170f034f987986891b5fc7a6455b5@192.168.1.84:5060' of Request 102: Match Found [Aug 10 15:45:24] DEBUG[17897] chan_sip.c: Destroying SIP dialog 01a170f034f987986891b5fc7a6455b5@192.168.1.84:5060 [Aug 10 15:45:24] VERBOSE[17897] chan_sip.c: Really destroying SIP dialog '01a170f034f987986891b5fc7a6455b5@192.168.1.84:5060' Method: OPTIONS [Aug 10 15:45:24] DEBUG[17897] chan_sip.c: Allocating new SIP dialog for 1464d6df38996aa306e966302ae64ba5@127.0.0.1:5060 - OPTIONS (No RTP) [Aug 10 15:45:24] DEBUG[17897] acl.c: For destination '192.168.1.102', our source address is '192.168.1.84'. [Aug 10 15:45:24] DEBUG[17897] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.1.84:5060 [Aug 10 15:45:24] DEBUG[17897] chan_sip.c: Initializing initreq for method OPTIONS - callid 7d5e641673ccf5700da02cf95b0f35f4@192.168.1.84:5060 [Aug 10 15:45:24] DEBUG[17897] chan_sip.c: Header 0 [ 57]: OPTIONS sip:2212@192.168.1.102:2048;line=z1zv7jsj SIP/2.0 [Aug 10 15:45:24] DEBUG[17897] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK309adca5;rport [Aug 10 15:45:24] DEBUG[17897] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 10 15:45:24] DEBUG[17897] chan_sip.c: Header 3 [ 59]: From: "asterisk" ;tag=as669a4556 [Aug 10 15:45:24] DEBUG[17897] chan_sip.c: Header 4 [ 47]: To: [Aug 10 15:45:24] DEBUG[17897] chan_sip.c: Header 5 [ 41]: Contact: [Aug 10 15:45:24] DEBUG[17897] chan_sip.c: Header 6 [ 59]: Call-ID: 7d5e641673ccf5700da02cf95b0f35f4@192.168.1.84:5060 [Aug 10 15:45:24] DEBUG[17897] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:45:24] DEBUG[17897] chan_sip.c: Header 8 [ 33]: User-Agent: Asterisk PBX 1.8.15.0 [Aug 10 15:45:24] DEBUG[17897] chan_sip.c: Header 9 [ 35]: Date: Fri, 10 Aug 2012 13:45:24 GMT [Aug 10 15:45:24] DEBUG[17897] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 10 15:45:24] DEBUG[17897] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 10 15:45:24] VERBOSE[17897] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.102:2048: OPTIONS sip:2212@192.168.1.102:2048;line=z1zv7jsj SIP/2.0 Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK309adca5;rport Max-Forwards: 70 From: "asterisk" ;tag=as669a4556 To: Contact: Call-ID: 7d5e641673ccf5700da02cf95b0f35f4@192.168.1.84:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.15.0 Date: Fri, 10 Aug 2012 13:45:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Aug 10 15:45:24] DEBUG[17897] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #283 [Aug 10 15:45:24] DEBUG[17897] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.1.102:2048 [Aug 10 15:45:24] VERBOSE[17897] chan_sip.c: <--- SIP read from UDP:192.168.1.102:2048 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK309adca5;rport=5060 From: "asterisk" ;tag=as669a4556 To: Call-ID: 7d5e641673ccf5700da02cf95b0f35f4@192.168.1.84:5060 CSeq: 102 OPTIONS Contact: ;reg-id=1 User-Agent: snom300/8.4.32 Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Content-Length: 0 <-------------> [Aug 10 15:45:24] DEBUG[17897] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 10 15:45:24] DEBUG[17897] chan_sip.c: Header 1 [ 68]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK309adca5;rport=5060 [Aug 10 15:45:24] DEBUG[17897] chan_sip.c: Header 2 [ 59]: From: "asterisk" ;tag=as669a4556 [Aug 10 15:45:24] DEBUG[17897] chan_sip.c: Header 3 [ 47]: To: [Aug 10 15:45:24] DEBUG[17897] chan_sip.c: Header 4 [ 59]: Call-ID: 7d5e641673ccf5700da02cf95b0f35f4@192.168.1.84:5060 [Aug 10 15:45:24] DEBUG[17897] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:45:24] DEBUG[17897] chan_sip.c: Header 6 [ 61]: Contact: ;reg-id=1 [Aug 10 15:45:24] DEBUG[17897] chan_sip.c: Header 7 [ 26]: User-Agent: snom300/8.4.32 [Aug 10 15:45:24] DEBUG[17897] chan_sip.c: Header 8 [ 19]: Accept-Language: en [Aug 10 15:45:24] DEBUG[17897] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [Aug 10 15:45:24] DEBUG[17897] chan_sip.c: Header 10 [ 96]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE [Aug 10 15:45:24] DEBUG[17897] chan_sip.c: Header 11 [ 42]: Allow-Events: talk, hold, refer, call-info [Aug 10 15:45:24] DEBUG[17897] chan_sip.c: Header 12 [ 47]: Supported: timer, 100rel, replaces, from-change [Aug 10 15:45:24] DEBUG[17897] chan_sip.c: Header 13 [ 17]: Content-Length: 0 [Aug 10 15:45:24] VERBOSE[17897] chan_sip.c: --- (14 headers 0 lines) --- [Aug 10 15:45:24] DEBUG[17897] chan_sip.c: = Looking for Call ID: 7d5e641673ccf5700da02cf95b0f35f4@192.168.1.84:5060 (Checking To) --From tag as669a4556 --To-tag [Aug 10 15:45:24] DEBUG[17897] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #283 [Aug 10 15:45:24] DEBUG[17897] chan_sip.c: Stopping retransmission on '7d5e641673ccf5700da02cf95b0f35f4@192.168.1.84:5060' of Request 102: Match Found [Aug 10 15:45:24] DEBUG[17897] chan_sip.c: Destroying SIP dialog 7d5e641673ccf5700da02cf95b0f35f4@192.168.1.84:5060 [Aug 10 15:45:24] VERBOSE[17897] chan_sip.c: Really destroying SIP dialog '7d5e641673ccf5700da02cf95b0f35f4@192.168.1.84:5060' Method: OPTIONS [Aug 10 15:45:29] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:45:29] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:45:34] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:45:34] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:45:39] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:45:39] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:45:44] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:45:44] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: Allocating new SIP dialog for 095d68c04165f24d50f2c96f087456f6@127.0.0.1:5060 - OPTIONS (No RTP) [Aug 10 15:45:45] DEBUG[17897] acl.c: For destination '192.168.1.106', our source address is '192.168.1.84'. [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.1.84:5060 [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: Initializing initreq for method OPTIONS - callid 6855eeac6e1481f76fb13fec0518e277@192.168.1.84:5060 [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: Header 0 [ 57]: OPTIONS sip:2219@192.168.1.106:2048;line=7g8ggsvj SIP/2.0 [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK786585de;rport [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: Header 3 [ 59]: From: "asterisk" ;tag=as0d71f9a7 [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: Header 4 [ 47]: To: [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: Header 5 [ 41]: Contact: [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: Header 6 [ 59]: Call-ID: 6855eeac6e1481f76fb13fec0518e277@192.168.1.84:5060 [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: Header 8 [ 33]: User-Agent: Asterisk PBX 1.8.15.0 [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: Header 9 [ 35]: Date: Fri, 10 Aug 2012 13:45:45 GMT [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 10 15:45:45] VERBOSE[17897] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.106:2048: OPTIONS sip:2219@192.168.1.106:2048;line=7g8ggsvj SIP/2.0 Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK786585de;rport Max-Forwards: 70 From: "asterisk" ;tag=as0d71f9a7 To: Contact: Call-ID: 6855eeac6e1481f76fb13fec0518e277@192.168.1.84:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.15.0 Date: Fri, 10 Aug 2012 13:45:45 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #286 [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.1.106:2048 [Aug 10 15:45:45] VERBOSE[17897] chan_sip.c: <--- SIP read from UDP:192.168.1.106:2048 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK786585de;rport=5060 From: "asterisk" ;tag=as0d71f9a7 To: ;tag=8mat5b2y0w Call-ID: 6855eeac6e1481f76fb13fec0518e277@192.168.1.84:5060 CSeq: 102 OPTIONS Contact: ;reg-id=1 User-Agent: snom300/8.7.3.10 Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: 100rel, replaces, from-change Content-Length: 0 <-------------> [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: Header 1 [ 68]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK786585de;rport=5060 [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: Header 2 [ 59]: From: "asterisk" ;tag=as0d71f9a7 [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: Header 3 [ 62]: To: ;tag=8mat5b2y0w [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: Header 4 [ 59]: Call-ID: 6855eeac6e1481f76fb13fec0518e277@192.168.1.84:5060 [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: Header 6 [ 61]: Contact: ;reg-id=1 [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: Header 7 [ 28]: User-Agent: snom300/8.7.3.10 [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: Header 8 [ 19]: Accept-Language: en [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: Header 10 [ 96]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: Header 11 [ 42]: Allow-Events: talk, hold, refer, call-info [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: Header 12 [ 40]: Supported: 100rel, replaces, from-change [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: Header 13 [ 17]: Content-Length: 0 [Aug 10 15:45:45] VERBOSE[17897] chan_sip.c: --- (14 headers 0 lines) --- [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: = Looking for Call ID: 6855eeac6e1481f76fb13fec0518e277@192.168.1.84:5060 (Checking To) --From tag as0d71f9a7 --To-tag 8mat5b2y0w [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #286 [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: Stopping retransmission on '6855eeac6e1481f76fb13fec0518e277@192.168.1.84:5060' of Request 102: Match Found [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: Destroying SIP dialog 6855eeac6e1481f76fb13fec0518e277@192.168.1.84:5060 [Aug 10 15:45:45] VERBOSE[17897] chan_sip.c: Really destroying SIP dialog '6855eeac6e1481f76fb13fec0518e277@192.168.1.84:5060' Method: OPTIONS [Aug 10 15:45:45] VERBOSE[17897] chan_sip.c: <--- SIP read from UDP:192.168.1.106:2048 ---> REGISTER sip:192.168.1.84 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.106:2048;branch=z9hG4bK-ilishnvw6bfq;rport From: "2219" ;tag=zy6fcr5r9j To: "2219" Call-ID: 50250e9b9684-8d3g9c7o4hy2 CSeq: 23 REGISTER Max-Forwards: 70 Contact: ;reg-id=1;q=1.0;+sip.instance="";audio;mobility="fixed";duplex="full";description="snom300";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO" User-Agent: snom300/8.7.3.10 Allow-Events: dialog X-Real-IP: 192.168.1.106 Supported: path, gruu Content-Length: 0 <-------------> [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: Header 0 [ 33]: REGISTER sip:192.168.1.84 SIP/2.0 [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: Header 1 [ 69]: Via: SIP/2.0/UDP 192.168.1.106:2048;branch=z9hG4bK-ilishnvw6bfq;rport [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: Header 2 [ 51]: From: "2219" ;tag=zy6fcr5r9j [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: Header 3 [ 34]: To: "2219" [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: Header 4 [ 34]: Call-ID: 50250e9b9684-8d3g9c7o4hy2 [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: Header 5 [ 17]: CSeq: 23 REGISTER [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: Header 7 [306]: Contact: ;reg-id=1;q=1.0;+sip.instance="";audio;mobility="fixed";duplex="full";description="snom300";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO" [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: Header 8 [ 28]: User-Agent: snom300/8.7.3.10 [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: Header 9 [ 20]: Allow-Events: dialog [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: Header 10 [ 24]: X-Real-IP: 192.168.1.106 [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: Header 11 [ 21]: Supported: path, gruu [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Aug 10 15:45:45] VERBOSE[17897] chan_sip.c: --- (13 headers 0 lines) --- [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: = Looking for Call ID: 50250e9b9684-8d3g9c7o4hy2 (Checking From) --From tag zy6fcr5r9j --To-tag [Aug 10 15:45:45] DEBUG[17897] acl.c: For destination '192.168.1.106', our source address is '192.168.1.84'. [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.1.84:5060 [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: Allocating new SIP dialog for 50250e9b9684-8d3g9c7o4hy2 - REGISTER (No RTP) [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: Initializing initreq for method REGISTER - callid 50250e9b9684-8d3g9c7o4hy2 [Aug 10 15:45:45] DEBUG[17897] netsock2.c: Splitting '192.168.1.106:2048' into... [Aug 10 15:45:45] DEBUG[17897] netsock2.c: ...host '192.168.1.106' and port '2048'. [Aug 10 15:45:45] VERBOSE[17897] chan_sip.c: Sending to 192.168.1.106:2048 (NAT) [Aug 10 15:45:45] DEBUG[17897] netsock2.c: Splitting '192.168.1.84' into... [Aug 10 15:45:45] DEBUG[17897] netsock2.c: ...host '192.168.1.84' and port ''. [Aug 10 15:45:45] VERBOSE[17897] chan_sip.c: <--- Transmitting (NAT) to 192.168.1.106:2048 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.106:2048;branch=z9hG4bK-ilishnvw6bfq;received=192.168.1.106;rport=2048 From: "2219" ;tag=zy6fcr5r9j To: "2219" ;tag=as28f91433 Call-ID: 50250e9b9684-8d3g9c7o4hy2 CSeq: 23 REGISTER Server: Asterisk PBX 1.8.15.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="11328e5e" Content-Length: 0 <------------> [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 192.168.1.106:2048 [Aug 10 15:45:45] VERBOSE[17897] chan_sip.c: Scheduling destruction of SIP dialog '50250e9b9684-8d3g9c7o4hy2' in 32000 ms (Method: REGISTER) [Aug 10 15:45:45] VERBOSE[17897] chan_sip.c: <--- SIP read from UDP:192.168.1.106:2048 ---> REGISTER sip:192.168.1.84 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.106:2048;branch=z9hG4bK-c5j1gl7ieo59;rport From: "2219" ;tag=zy6fcr5r9j To: "2219" Call-ID: 50250e9b9684-8d3g9c7o4hy2 CSeq: 24 REGISTER Max-Forwards: 70 Contact: ;reg-id=1;q=1.0;+sip.instance="";audio;mobility="fixed";duplex="full";description="snom300";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO" User-Agent: snom300/8.7.3.10 Allow-Events: dialog X-Real-IP: 192.168.1.106 Supported: path, gruu Authorization: Digest username="2219",realm="asterisk",nonce="11328e5e",uri="sip:192.168.1.84",response="52571cb0fe4d55ac9cf66b9258fe2eef",algorithm=MD5 Content-Length: 0 <-------------> [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: Header 0 [ 33]: REGISTER sip:192.168.1.84 SIP/2.0 [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: Header 1 [ 69]: Via: SIP/2.0/UDP 192.168.1.106:2048;branch=z9hG4bK-c5j1gl7ieo59;rport [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: Header 2 [ 51]: From: "2219" ;tag=zy6fcr5r9j [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: Header 3 [ 34]: To: "2219" [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: Header 4 [ 34]: Call-ID: 50250e9b9684-8d3g9c7o4hy2 [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: Header 5 [ 17]: CSeq: 24 REGISTER [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: Header 7 [306]: Contact: ;reg-id=1;q=1.0;+sip.instance="";audio;mobility="fixed";duplex="full";description="snom300";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO" [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: Header 8 [ 28]: User-Agent: snom300/8.7.3.10 [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: Header 9 [ 20]: Allow-Events: dialog [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: Header 10 [ 24]: X-Real-IP: 192.168.1.106 [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: Header 11 [ 21]: Supported: path, gruu [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: Header 12 [152]: Authorization: Digest username="2219",realm="asterisk",nonce="11328e5e",uri="sip:192.168.1.84",response="52571cb0fe4d55ac9cf66b9258fe2eef",algorithm=MD5 [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: Header 13 [ 17]: Content-Length: 0 [Aug 10 15:45:45] VERBOSE[17897] chan_sip.c: --- (14 headers 0 lines) --- [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: = Looking for Call ID: 50250e9b9684-8d3g9c7o4hy2 (Checking From) --From tag zy6fcr5r9j --To-tag [Aug 10 15:45:45] DEBUG[17897] netsock2.c: Splitting '192.168.1.84' into... [Aug 10 15:45:45] DEBUG[17897] netsock2.c: ...host '192.168.1.84' and port ''. [Aug 10 15:45:45] DEBUG[17897] netsock2.c: Splitting '192.168.1.84' into... [Aug 10 15:45:45] DEBUG[17897] netsock2.c: ...host '192.168.1.84' and port ''. [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: Initializing initreq for method REGISTER - callid 50250e9b9684-8d3g9c7o4hy2 [Aug 10 15:45:45] DEBUG[17897] netsock2.c: Splitting '192.168.1.106:2048' into... [Aug 10 15:45:45] DEBUG[17897] netsock2.c: ...host '192.168.1.106' and port '2048'. [Aug 10 15:45:45] VERBOSE[17897] chan_sip.c: Sending to 192.168.1.106:2048 (NAT) [Aug 10 15:45:45] DEBUG[17897] netsock2.c: Splitting '192.168.1.84' into... [Aug 10 15:45:45] DEBUG[17897] netsock2.c: ...host '192.168.1.84' and port ''. [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: Store REGISTER's src-IP:port for call routing. [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: Allocating new SIP dialog for 7451f72117b17ebe705e3f671e358e38@127.0.0.1:5060 - OPTIONS (No RTP) [Aug 10 15:45:45] DEBUG[17897] acl.c: For destination '192.168.1.106', our source address is '192.168.1.84'. [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.1.84:5060 [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: Initializing initreq for method OPTIONS - callid 03ed572b27fa2ba55721e50a5ed60a18@192.168.1.84:5060 [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: Header 0 [ 57]: OPTIONS sip:2219@192.168.1.106:2048;line=7g8ggsvj SIP/2.0 [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK16b437f8;rport [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: Header 3 [ 59]: From: "asterisk" ;tag=as6b3519dc [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: Header 4 [ 47]: To: [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: Header 5 [ 41]: Contact: [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: Header 6 [ 59]: Call-ID: 03ed572b27fa2ba55721e50a5ed60a18@192.168.1.84:5060 [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: Header 8 [ 33]: User-Agent: Asterisk PBX 1.8.15.0 [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: Header 9 [ 35]: Date: Fri, 10 Aug 2012 13:45:45 GMT [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 10 15:45:45] VERBOSE[17897] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.106:2048: OPTIONS sip:2219@192.168.1.106:2048;line=7g8ggsvj SIP/2.0 Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK16b437f8;rport Max-Forwards: 70 From: "asterisk" ;tag=as6b3519dc To: Contact: Call-ID: 03ed572b27fa2ba55721e50a5ed60a18@192.168.1.84:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.15.0 Date: Fri, 10 Aug 2012 13:45:45 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #291 [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.1.106:2048 [Aug 10 15:45:45] VERBOSE[17897] chan_sip.c: <--- Transmitting (NAT) to 192.168.1.106:2048 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.106:2048;branch=z9hG4bK-c5j1gl7ieo59;received=192.168.1.106;rport=2048 From: "2219" ;tag=zy6fcr5r9j To: "2219" ;tag=as28f91433 Call-ID: 50250e9b9684-8d3g9c7o4hy2 CSeq: 24 REGISTER Server: Asterisk PBX 1.8.15.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Expires: 120 Contact: ;expires=120 Date: Fri, 10 Aug 2012 13:45:45 GMT Content-Length: 0 <------------> [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.1.106:2048 [Aug 10 15:45:45] VERBOSE[17897] chan_sip.c: Scheduling destruction of SIP dialog '50250e9b9684-8d3g9c7o4hy2' in 32000 ms (Method: REGISTER) [Aug 10 15:45:45] DEBUG[17874] devicestate.c: No provider found, checking channel drivers for SIP - 2219 [Aug 10 15:45:45] DEBUG[17874] chan_sip.c: Checking device state for peer 2219 [Aug 10 15:45:45] DEBUG[17874] devicestate.c: Changing state for SIP/2219 - state 1 (Not in use) [Aug 10 15:45:45] DEBUG[17874] devicestate.c: device 'SIP/2219' state '1' [Aug 10 15:45:45] DEBUG[17907] app_queue.c: Device 'SIP/2219' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Aug 10 15:45:45] VERBOSE[17897] chan_sip.c: <--- SIP read from UDP:192.168.1.106:2048 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK16b437f8;rport=5060 From: "asterisk" ;tag=as6b3519dc To: ;tag=gv4rmeclo9 Call-ID: 03ed572b27fa2ba55721e50a5ed60a18@192.168.1.84:5060 CSeq: 102 OPTIONS Contact: ;reg-id=1 User-Agent: snom300/8.7.3.10 Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: 100rel, replaces, from-change Content-Length: 0 <-------------> [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: Header 1 [ 68]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK16b437f8;rport=5060 [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: Header 2 [ 59]: From: "asterisk" ;tag=as6b3519dc [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: Header 3 [ 62]: To: ;tag=gv4rmeclo9 [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: Header 4 [ 59]: Call-ID: 03ed572b27fa2ba55721e50a5ed60a18@192.168.1.84:5060 [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: Header 6 [ 61]: Contact: ;reg-id=1 [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: Header 7 [ 28]: User-Agent: snom300/8.7.3.10 [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: Header 8 [ 19]: Accept-Language: en [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: Header 10 [ 96]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: Header 11 [ 42]: Allow-Events: talk, hold, refer, call-info [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: Header 12 [ 40]: Supported: 100rel, replaces, from-change [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: Header 13 [ 17]: Content-Length: 0 [Aug 10 15:45:45] VERBOSE[17897] chan_sip.c: --- (14 headers 0 lines) --- [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: = Looking for Call ID: 03ed572b27fa2ba55721e50a5ed60a18@192.168.1.84:5060 (Checking To) --From tag as6b3519dc --To-tag gv4rmeclo9 [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #291 [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: Stopping retransmission on '03ed572b27fa2ba55721e50a5ed60a18@192.168.1.84:5060' of Request 102: Match Found [Aug 10 15:45:45] DEBUG[17897] chan_sip.c: Destroying SIP dialog 03ed572b27fa2ba55721e50a5ed60a18@192.168.1.84:5060 [Aug 10 15:45:45] VERBOSE[17897] chan_sip.c: Really destroying SIP dialog '03ed572b27fa2ba55721e50a5ed60a18@192.168.1.84:5060' Method: OPTIONS [Aug 10 15:45:49] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:45:49] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:45:54] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:45:54] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:45:59] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:45:59] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:46:04] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:46:04] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:46:09] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:46:09] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:46:14] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:46:14] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:46:17] DEBUG[17897] chan_sip.c: Auto destroying SIP dialog '50250e9b9684-8d3g9c7o4hy2' [Aug 10 15:46:17] DEBUG[17897] chan_sip.c: Destroying SIP dialog 50250e9b9684-8d3g9c7o4hy2 [Aug 10 15:46:17] VERBOSE[17897] chan_sip.c: Really destroying SIP dialog '50250e9b9684-8d3g9c7o4hy2' Method: REGISTER [Aug 10 15:46:19] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:46:19] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:46:24] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:46:24] DEBUG[17897] chan_sip.c: Allocating new SIP dialog for 13e1d77f5fc26fd230bbb4cc4f94aa55@127.0.0.1:5060 - OPTIONS (No RTP) [Aug 10 15:46:24] DEBUG[17897] acl.c: For destination '192.168.2.210', our source address is '192.168.1.84'. [Aug 10 15:46:24] DEBUG[17897] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.1.84:5060 [Aug 10 15:46:24] DEBUG[17897] chan_sip.c: Initializing initreq for method OPTIONS - callid 268d36997a25b3912f4deccc542cde9b@192.168.1.84:5060 [Aug 10 15:46:24] DEBUG[17897] chan_sip.c: Header 0 [ 43]: OPTIONS sip:2210@192.168.2.210:2048 SIP/2.0 [Aug 10 15:46:24] DEBUG[17897] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK2dcc97a9;rport [Aug 10 15:46:24] DEBUG[17897] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 10 15:46:24] DEBUG[17897] chan_sip.c: Header 3 [ 59]: From: "asterisk" ;tag=as20025e6d [Aug 10 15:46:24] DEBUG[17897] chan_sip.c: Header 4 [ 33]: To: [Aug 10 15:46:24] DEBUG[17897] chan_sip.c: Header 5 [ 41]: Contact: [Aug 10 15:46:24] DEBUG[17897] chan_sip.c: Header 6 [ 59]: Call-ID: 268d36997a25b3912f4deccc542cde9b@192.168.1.84:5060 [Aug 10 15:46:24] DEBUG[17897] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:46:24] DEBUG[17897] chan_sip.c: Header 8 [ 33]: User-Agent: Asterisk PBX 1.8.15.0 [Aug 10 15:46:24] DEBUG[17897] chan_sip.c: Header 9 [ 35]: Date: Fri, 10 Aug 2012 13:46:24 GMT [Aug 10 15:46:24] DEBUG[17897] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 10 15:46:24] DEBUG[17897] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 10 15:46:24] VERBOSE[17897] chan_sip.c: Reliably Transmitting (NAT) to 192.168.2.210:2048: OPTIONS sip:2210@192.168.2.210:2048 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK2dcc97a9;rport Max-Forwards: 70 From: "asterisk" ;tag=as20025e6d To: Contact: Call-ID: 268d36997a25b3912f4deccc542cde9b@192.168.1.84:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.15.0 Date: Fri, 10 Aug 2012 13:46:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Aug 10 15:46:24] DEBUG[17897] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #295 [Aug 10 15:46:24] DEBUG[17897] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.2.210:2048 [Aug 10 15:46:24] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:46:24] VERBOSE[17897] chan_sip.c: <--- SIP read from UDP:192.168.2.210:2048 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK2dcc97a9;rport=5060 From: "asterisk" ;tag=as20025e6d To: Call-ID: 268d36997a25b3912f4deccc542cde9b@192.168.1.84:5060 CSeq: 102 OPTIONS Contact: ;reg-id=1 User-Agent: snom300/8.4.32 Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: 100rel, replaces, from-change Content-Length: 0 <-------------> [Aug 10 15:46:24] DEBUG[17897] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 10 15:46:24] DEBUG[17897] chan_sip.c: Header 1 [ 68]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK2dcc97a9;rport=5060 [Aug 10 15:46:24] DEBUG[17897] chan_sip.c: Header 2 [ 59]: From: "asterisk" ;tag=as20025e6d [Aug 10 15:46:24] DEBUG[17897] chan_sip.c: Header 3 [ 33]: To: [Aug 10 15:46:24] DEBUG[17897] chan_sip.c: Header 4 [ 59]: Call-ID: 268d36997a25b3912f4deccc542cde9b@192.168.1.84:5060 [Aug 10 15:46:24] DEBUG[17897] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:46:24] DEBUG[17897] chan_sip.c: Header 6 [ 47]: Contact: ;reg-id=1 [Aug 10 15:46:24] DEBUG[17897] chan_sip.c: Header 7 [ 26]: User-Agent: snom300/8.4.32 [Aug 10 15:46:24] DEBUG[17897] chan_sip.c: Header 8 [ 19]: Accept-Language: en [Aug 10 15:46:24] DEBUG[17897] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [Aug 10 15:46:24] DEBUG[17897] chan_sip.c: Header 10 [ 96]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE [Aug 10 15:46:24] DEBUG[17897] chan_sip.c: Header 11 [ 42]: Allow-Events: talk, hold, refer, call-info [Aug 10 15:46:24] DEBUG[17897] chan_sip.c: Header 12 [ 40]: Supported: 100rel, replaces, from-change [Aug 10 15:46:24] DEBUG[17897] chan_sip.c: Header 13 [ 17]: Content-Length: 0 [Aug 10 15:46:24] VERBOSE[17897] chan_sip.c: --- (14 headers 0 lines) --- [Aug 10 15:46:24] DEBUG[17897] chan_sip.c: = Looking for Call ID: 268d36997a25b3912f4deccc542cde9b@192.168.1.84:5060 (Checking To) --From tag as20025e6d --To-tag [Aug 10 15:46:24] DEBUG[17897] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #295 [Aug 10 15:46:24] DEBUG[17897] chan_sip.c: Stopping retransmission on '268d36997a25b3912f4deccc542cde9b@192.168.1.84:5060' of Request 102: Match Found [Aug 10 15:46:24] DEBUG[17897] chan_sip.c: Destroying SIP dialog 268d36997a25b3912f4deccc542cde9b@192.168.1.84:5060 [Aug 10 15:46:24] VERBOSE[17897] chan_sip.c: Really destroying SIP dialog '268d36997a25b3912f4deccc542cde9b@192.168.1.84:5060' Method: OPTIONS [Aug 10 15:46:24] DEBUG[17897] chan_sip.c: Allocating new SIP dialog for 5ec6d8ba76ad275c6efd91c75e6914f5@127.0.0.1:5060 - OPTIONS (No RTP) [Aug 10 15:46:24] DEBUG[17897] acl.c: For destination '192.168.1.102', our source address is '192.168.1.84'. [Aug 10 15:46:24] DEBUG[17897] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.1.84:5060 [Aug 10 15:46:24] DEBUG[17897] chan_sip.c: Initializing initreq for method OPTIONS - callid 356b5a1779c013a03c5a68e50dc11c2c@192.168.1.84:5060 [Aug 10 15:46:24] DEBUG[17897] chan_sip.c: Header 0 [ 57]: OPTIONS sip:2212@192.168.1.102:2048;line=z1zv7jsj SIP/2.0 [Aug 10 15:46:24] DEBUG[17897] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK46d5db54;rport [Aug 10 15:46:24] DEBUG[17897] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 10 15:46:24] DEBUG[17897] chan_sip.c: Header 3 [ 59]: From: "asterisk" ;tag=as132413ae [Aug 10 15:46:24] DEBUG[17897] chan_sip.c: Header 4 [ 47]: To: [Aug 10 15:46:24] DEBUG[17897] chan_sip.c: Header 5 [ 41]: Contact: [Aug 10 15:46:24] DEBUG[17897] chan_sip.c: Header 6 [ 59]: Call-ID: 356b5a1779c013a03c5a68e50dc11c2c@192.168.1.84:5060 [Aug 10 15:46:24] DEBUG[17897] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:46:24] DEBUG[17897] chan_sip.c: Header 8 [ 33]: User-Agent: Asterisk PBX 1.8.15.0 [Aug 10 15:46:24] DEBUG[17897] chan_sip.c: Header 9 [ 35]: Date: Fri, 10 Aug 2012 13:46:24 GMT [Aug 10 15:46:24] DEBUG[17897] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 10 15:46:24] DEBUG[17897] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 10 15:46:24] VERBOSE[17897] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.102:2048: OPTIONS sip:2212@192.168.1.102:2048;line=z1zv7jsj SIP/2.0 Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK46d5db54;rport Max-Forwards: 70 From: "asterisk" ;tag=as132413ae To: Contact: Call-ID: 356b5a1779c013a03c5a68e50dc11c2c@192.168.1.84:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.15.0 Date: Fri, 10 Aug 2012 13:46:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Aug 10 15:46:24] DEBUG[17897] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #298 [Aug 10 15:46:24] DEBUG[17897] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.1.102:2048 [Aug 10 15:46:24] VERBOSE[17897] chan_sip.c: <--- SIP read from UDP:192.168.1.102:2048 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK46d5db54;rport=5060 From: "asterisk" ;tag=as132413ae To: Call-ID: 356b5a1779c013a03c5a68e50dc11c2c@192.168.1.84:5060 CSeq: 102 OPTIONS Contact: ;reg-id=1 User-Agent: snom300/8.4.32 Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Content-Length: 0 <-------------> [Aug 10 15:46:24] DEBUG[17897] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 10 15:46:24] DEBUG[17897] chan_sip.c: Header 1 [ 68]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK46d5db54;rport=5060 [Aug 10 15:46:24] DEBUG[17897] chan_sip.c: Header 2 [ 59]: From: "asterisk" ;tag=as132413ae [Aug 10 15:46:24] DEBUG[17897] chan_sip.c: Header 3 [ 47]: To: [Aug 10 15:46:24] DEBUG[17897] chan_sip.c: Header 4 [ 59]: Call-ID: 356b5a1779c013a03c5a68e50dc11c2c@192.168.1.84:5060 [Aug 10 15:46:24] DEBUG[17897] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:46:24] DEBUG[17897] chan_sip.c: Header 6 [ 61]: Contact: ;reg-id=1 [Aug 10 15:46:24] DEBUG[17897] chan_sip.c: Header 7 [ 26]: User-Agent: snom300/8.4.32 [Aug 10 15:46:24] DEBUG[17897] chan_sip.c: Header 8 [ 19]: Accept-Language: en [Aug 10 15:46:24] DEBUG[17897] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [Aug 10 15:46:24] DEBUG[17897] chan_sip.c: Header 10 [ 96]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE [Aug 10 15:46:24] DEBUG[17897] chan_sip.c: Header 11 [ 42]: Allow-Events: talk, hold, refer, call-info [Aug 10 15:46:24] DEBUG[17897] chan_sip.c: Header 12 [ 47]: Supported: timer, 100rel, replaces, from-change [Aug 10 15:46:24] DEBUG[17897] chan_sip.c: Header 13 [ 17]: Content-Length: 0 [Aug 10 15:46:24] VERBOSE[17897] chan_sip.c: --- (14 headers 0 lines) --- [Aug 10 15:46:24] DEBUG[17897] chan_sip.c: = Looking for Call ID: 356b5a1779c013a03c5a68e50dc11c2c@192.168.1.84:5060 (Checking To) --From tag as132413ae --To-tag [Aug 10 15:46:24] DEBUG[17897] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #298 [Aug 10 15:46:24] DEBUG[17897] chan_sip.c: Stopping retransmission on '356b5a1779c013a03c5a68e50dc11c2c@192.168.1.84:5060' of Request 102: Match Found [Aug 10 15:46:24] DEBUG[17897] chan_sip.c: Destroying SIP dialog 356b5a1779c013a03c5a68e50dc11c2c@192.168.1.84:5060 [Aug 10 15:46:24] VERBOSE[17897] chan_sip.c: Really destroying SIP dialog '356b5a1779c013a03c5a68e50dc11c2c@192.168.1.84:5060' Method: OPTIONS [Aug 10 15:46:29] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:46:29] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:46:34] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:46:34] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:46:39] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:46:39] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:46:44] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:46:44] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: Allocating new SIP dialog for 0053446869452b6a5baedfdb7cd94b64@127.0.0.1:5060 - OPTIONS (No RTP) [Aug 10 15:46:45] DEBUG[17897] acl.c: For destination '192.168.1.106', our source address is '192.168.1.84'. [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.1.84:5060 [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: Initializing initreq for method OPTIONS - callid 454bcdab6872a41a3fab768b3d6ece5a@192.168.1.84:5060 [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: Header 0 [ 57]: OPTIONS sip:2219@192.168.1.106:2048;line=7g8ggsvj SIP/2.0 [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK6c2d9630;rport [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: Header 3 [ 59]: From: "asterisk" ;tag=as2a2eb23b [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: Header 4 [ 47]: To: [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: Header 5 [ 41]: Contact: [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: Header 6 [ 59]: Call-ID: 454bcdab6872a41a3fab768b3d6ece5a@192.168.1.84:5060 [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: Header 8 [ 33]: User-Agent: Asterisk PBX 1.8.15.0 [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: Header 9 [ 35]: Date: Fri, 10 Aug 2012 13:46:45 GMT [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 10 15:46:45] VERBOSE[17897] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.106:2048: OPTIONS sip:2219@192.168.1.106:2048;line=7g8ggsvj SIP/2.0 Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK6c2d9630;rport Max-Forwards: 70 From: "asterisk" ;tag=as2a2eb23b To: Contact: Call-ID: 454bcdab6872a41a3fab768b3d6ece5a@192.168.1.84:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.15.0 Date: Fri, 10 Aug 2012 13:46:45 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #301 [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.1.106:2048 [Aug 10 15:46:45] VERBOSE[17897] chan_sip.c: <--- SIP read from UDP:192.168.1.106:2048 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK6c2d9630;rport=5060 From: "asterisk" ;tag=as2a2eb23b To: ;tag=dwxmanokv2 Call-ID: 454bcdab6872a41a3fab768b3d6ece5a@192.168.1.84:5060 CSeq: 102 OPTIONS Contact: ;reg-id=1 User-Agent: snom300/8.7.3.10 Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: 100rel, replaces, from-change Content-Length: 0 <-------------> [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: Header 1 [ 68]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK6c2d9630;rport=5060 [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: Header 2 [ 59]: From: "asterisk" ;tag=as2a2eb23b [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: Header 3 [ 62]: To: ;tag=dwxmanokv2 [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: Header 4 [ 59]: Call-ID: 454bcdab6872a41a3fab768b3d6ece5a@192.168.1.84:5060 [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: Header 6 [ 61]: Contact: ;reg-id=1 [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: Header 7 [ 28]: User-Agent: snom300/8.7.3.10 [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: Header 8 [ 19]: Accept-Language: en [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: Header 10 [ 96]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: Header 11 [ 42]: Allow-Events: talk, hold, refer, call-info [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: Header 12 [ 40]: Supported: 100rel, replaces, from-change [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: Header 13 [ 17]: Content-Length: 0 [Aug 10 15:46:45] VERBOSE[17897] chan_sip.c: --- (14 headers 0 lines) --- [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: = Looking for Call ID: 454bcdab6872a41a3fab768b3d6ece5a@192.168.1.84:5060 (Checking To) --From tag as2a2eb23b --To-tag dwxmanokv2 [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #301 [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: Stopping retransmission on '454bcdab6872a41a3fab768b3d6ece5a@192.168.1.84:5060' of Request 102: Match Found [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: Destroying SIP dialog 454bcdab6872a41a3fab768b3d6ece5a@192.168.1.84:5060 [Aug 10 15:46:45] VERBOSE[17897] chan_sip.c: Really destroying SIP dialog '454bcdab6872a41a3fab768b3d6ece5a@192.168.1.84:5060' Method: OPTIONS [Aug 10 15:46:45] VERBOSE[17897] chan_sip.c: <--- SIP read from UDP:192.168.1.106:2048 ---> REGISTER sip:192.168.1.84 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.106:2048;branch=z9hG4bK-r8ldt67crcmg;rport From: "2219" ;tag=1bw62z0e1o To: "2219" Call-ID: 50250e9b9684-8d3g9c7o4hy2 CSeq: 25 REGISTER Max-Forwards: 70 Contact: ;reg-id=1;q=1.0;+sip.instance="";audio;mobility="fixed";duplex="full";description="snom300";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO" User-Agent: snom300/8.7.3.10 Allow-Events: dialog X-Real-IP: 192.168.1.106 Supported: path, gruu Content-Length: 0 <-------------> [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: Header 0 [ 33]: REGISTER sip:192.168.1.84 SIP/2.0 [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: Header 1 [ 69]: Via: SIP/2.0/UDP 192.168.1.106:2048;branch=z9hG4bK-r8ldt67crcmg;rport [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: Header 2 [ 51]: From: "2219" ;tag=1bw62z0e1o [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: Header 3 [ 34]: To: "2219" [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: Header 4 [ 34]: Call-ID: 50250e9b9684-8d3g9c7o4hy2 [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: Header 5 [ 17]: CSeq: 25 REGISTER [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: Header 7 [306]: Contact: ;reg-id=1;q=1.0;+sip.instance="";audio;mobility="fixed";duplex="full";description="snom300";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO" [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: Header 8 [ 28]: User-Agent: snom300/8.7.3.10 [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: Header 9 [ 20]: Allow-Events: dialog [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: Header 10 [ 24]: X-Real-IP: 192.168.1.106 [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: Header 11 [ 21]: Supported: path, gruu [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Aug 10 15:46:45] VERBOSE[17897] chan_sip.c: --- (13 headers 0 lines) --- [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: = Looking for Call ID: 50250e9b9684-8d3g9c7o4hy2 (Checking From) --From tag 1bw62z0e1o --To-tag [Aug 10 15:46:45] DEBUG[17897] acl.c: For destination '192.168.1.106', our source address is '192.168.1.84'. [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.1.84:5060 [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: Allocating new SIP dialog for 50250e9b9684-8d3g9c7o4hy2 - REGISTER (No RTP) [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: Initializing initreq for method REGISTER - callid 50250e9b9684-8d3g9c7o4hy2 [Aug 10 15:46:45] DEBUG[17897] netsock2.c: Splitting '192.168.1.106:2048' into... [Aug 10 15:46:45] DEBUG[17897] netsock2.c: ...host '192.168.1.106' and port '2048'. [Aug 10 15:46:45] VERBOSE[17897] chan_sip.c: Sending to 192.168.1.106:2048 (NAT) [Aug 10 15:46:45] DEBUG[17897] netsock2.c: Splitting '192.168.1.84' into... [Aug 10 15:46:45] DEBUG[17897] netsock2.c: ...host '192.168.1.84' and port ''. [Aug 10 15:46:45] VERBOSE[17897] chan_sip.c: <--- Transmitting (NAT) to 192.168.1.106:2048 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.106:2048;branch=z9hG4bK-r8ldt67crcmg;received=192.168.1.106;rport=2048 From: "2219" ;tag=1bw62z0e1o To: "2219" ;tag=as51ad9029 Call-ID: 50250e9b9684-8d3g9c7o4hy2 CSeq: 25 REGISTER Server: Asterisk PBX 1.8.15.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="29f23e68" Content-Length: 0 <------------> [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 192.168.1.106:2048 [Aug 10 15:46:45] VERBOSE[17897] chan_sip.c: Scheduling destruction of SIP dialog '50250e9b9684-8d3g9c7o4hy2' in 32000 ms (Method: REGISTER) [Aug 10 15:46:45] VERBOSE[17897] chan_sip.c: <--- SIP read from UDP:192.168.1.106:2048 ---> REGISTER sip:192.168.1.84 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.106:2048;branch=z9hG4bK-i0of8znud0kk;rport From: "2219" ;tag=1bw62z0e1o To: "2219" Call-ID: 50250e9b9684-8d3g9c7o4hy2 CSeq: 26 REGISTER Max-Forwards: 70 Contact: ;reg-id=1;q=1.0;+sip.instance="";audio;mobility="fixed";duplex="full";description="snom300";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO" User-Agent: snom300/8.7.3.10 Allow-Events: dialog X-Real-IP: 192.168.1.106 Supported: path, gruu Authorization: Digest username="2219",realm="asterisk",nonce="29f23e68",uri="sip:192.168.1.84",response="8c07e7c520e95b042c3e010be4f54e43",algorithm=MD5 Content-Length: 0 <-------------> [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: Header 0 [ 33]: REGISTER sip:192.168.1.84 SIP/2.0 [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: Header 1 [ 69]: Via: SIP/2.0/UDP 192.168.1.106:2048;branch=z9hG4bK-i0of8znud0kk;rport [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: Header 2 [ 51]: From: "2219" ;tag=1bw62z0e1o [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: Header 3 [ 34]: To: "2219" [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: Header 4 [ 34]: Call-ID: 50250e9b9684-8d3g9c7o4hy2 [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: Header 5 [ 17]: CSeq: 26 REGISTER [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: Header 7 [306]: Contact: ;reg-id=1;q=1.0;+sip.instance="";audio;mobility="fixed";duplex="full";description="snom300";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO" [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: Header 8 [ 28]: User-Agent: snom300/8.7.3.10 [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: Header 9 [ 20]: Allow-Events: dialog [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: Header 10 [ 24]: X-Real-IP: 192.168.1.106 [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: Header 11 [ 21]: Supported: path, gruu [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: Header 12 [152]: Authorization: Digest username="2219",realm="asterisk",nonce="29f23e68",uri="sip:192.168.1.84",response="8c07e7c520e95b042c3e010be4f54e43",algorithm=MD5 [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: Header 13 [ 17]: Content-Length: 0 [Aug 10 15:46:45] VERBOSE[17897] chan_sip.c: --- (14 headers 0 lines) --- [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: = Looking for Call ID: 50250e9b9684-8d3g9c7o4hy2 (Checking From) --From tag 1bw62z0e1o --To-tag [Aug 10 15:46:45] DEBUG[17897] netsock2.c: Splitting '192.168.1.84' into... [Aug 10 15:46:45] DEBUG[17897] netsock2.c: ...host '192.168.1.84' and port ''. [Aug 10 15:46:45] DEBUG[17897] netsock2.c: Splitting '192.168.1.84' into... [Aug 10 15:46:45] DEBUG[17897] netsock2.c: ...host '192.168.1.84' and port ''. [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: Initializing initreq for method REGISTER - callid 50250e9b9684-8d3g9c7o4hy2 [Aug 10 15:46:45] DEBUG[17897] netsock2.c: Splitting '192.168.1.106:2048' into... [Aug 10 15:46:45] DEBUG[17897] netsock2.c: ...host '192.168.1.106' and port '2048'. [Aug 10 15:46:45] VERBOSE[17897] chan_sip.c: Sending to 192.168.1.106:2048 (NAT) [Aug 10 15:46:45] DEBUG[17897] netsock2.c: Splitting '192.168.1.84' into... [Aug 10 15:46:45] DEBUG[17897] netsock2.c: ...host '192.168.1.84' and port ''. [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: Store REGISTER's src-IP:port for call routing. [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: Allocating new SIP dialog for 4d0e2c1569b60e894e8fcd5a16481f8d@127.0.0.1:5060 - OPTIONS (No RTP) [Aug 10 15:46:45] DEBUG[17897] acl.c: For destination '192.168.1.106', our source address is '192.168.1.84'. [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.1.84:5060 [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: Initializing initreq for method OPTIONS - callid 354b26f06bb189d8105e6c3821471a85@192.168.1.84:5060 [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: Header 0 [ 57]: OPTIONS sip:2219@192.168.1.106:2048;line=7g8ggsvj SIP/2.0 [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK14d2ff1d;rport [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: Header 3 [ 59]: From: "asterisk" ;tag=as5546249a [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: Header 4 [ 47]: To: [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: Header 5 [ 41]: Contact: [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: Header 6 [ 59]: Call-ID: 354b26f06bb189d8105e6c3821471a85@192.168.1.84:5060 [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: Header 8 [ 33]: User-Agent: Asterisk PBX 1.8.15.0 [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: Header 9 [ 35]: Date: Fri, 10 Aug 2012 13:46:45 GMT [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 10 15:46:45] VERBOSE[17897] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.106:2048: OPTIONS sip:2219@192.168.1.106:2048;line=7g8ggsvj SIP/2.0 Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK14d2ff1d;rport Max-Forwards: 70 From: "asterisk" ;tag=as5546249a To: Contact: Call-ID: 354b26f06bb189d8105e6c3821471a85@192.168.1.84:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.15.0 Date: Fri, 10 Aug 2012 13:46:45 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #306 [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.1.106:2048 [Aug 10 15:46:45] VERBOSE[17897] chan_sip.c: <--- Transmitting (NAT) to 192.168.1.106:2048 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.106:2048;branch=z9hG4bK-i0of8znud0kk;received=192.168.1.106;rport=2048 From: "2219" ;tag=1bw62z0e1o To: "2219" ;tag=as51ad9029 Call-ID: 50250e9b9684-8d3g9c7o4hy2 CSeq: 26 REGISTER Server: Asterisk PBX 1.8.15.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Expires: 120 Contact: ;expires=120 Date: Fri, 10 Aug 2012 13:46:45 GMT Content-Length: 0 <------------> [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.1.106:2048 [Aug 10 15:46:45] VERBOSE[17897] chan_sip.c: Scheduling destruction of SIP dialog '50250e9b9684-8d3g9c7o4hy2' in 32000 ms (Method: REGISTER) [Aug 10 15:46:45] DEBUG[17874] devicestate.c: No provider found, checking channel drivers for SIP - 2219 [Aug 10 15:46:45] DEBUG[17874] chan_sip.c: Checking device state for peer 2219 [Aug 10 15:46:45] DEBUG[17874] devicestate.c: Changing state for SIP/2219 - state 1 (Not in use) [Aug 10 15:46:45] DEBUG[17874] devicestate.c: device 'SIP/2219' state '1' [Aug 10 15:46:45] DEBUG[17907] app_queue.c: Device 'SIP/2219' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Aug 10 15:46:45] VERBOSE[17897] chan_sip.c: <--- SIP read from UDP:192.168.1.106:2048 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK14d2ff1d;rport=5060 From: "asterisk" ;tag=as5546249a To: ;tag=kr0jg2dqyb Call-ID: 354b26f06bb189d8105e6c3821471a85@192.168.1.84:5060 CSeq: 102 OPTIONS Contact: ;reg-id=1 User-Agent: snom300/8.7.3.10 Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: 100rel, replaces, from-change Content-Length: 0 <-------------> [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: Header 1 [ 68]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK14d2ff1d;rport=5060 [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: Header 2 [ 59]: From: "asterisk" ;tag=as5546249a [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: Header 3 [ 62]: To: ;tag=kr0jg2dqyb [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: Header 4 [ 59]: Call-ID: 354b26f06bb189d8105e6c3821471a85@192.168.1.84:5060 [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: Header 6 [ 61]: Contact: ;reg-id=1 [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: Header 7 [ 28]: User-Agent: snom300/8.7.3.10 [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: Header 8 [ 19]: Accept-Language: en [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: Header 10 [ 96]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: Header 11 [ 42]: Allow-Events: talk, hold, refer, call-info [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: Header 12 [ 40]: Supported: 100rel, replaces, from-change [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: Header 13 [ 17]: Content-Length: 0 [Aug 10 15:46:45] VERBOSE[17897] chan_sip.c: --- (14 headers 0 lines) --- [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: = Looking for Call ID: 354b26f06bb189d8105e6c3821471a85@192.168.1.84:5060 (Checking To) --From tag as5546249a --To-tag kr0jg2dqyb [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #306 [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: Stopping retransmission on '354b26f06bb189d8105e6c3821471a85@192.168.1.84:5060' of Request 102: Match Found [Aug 10 15:46:45] DEBUG[17897] chan_sip.c: Destroying SIP dialog 354b26f06bb189d8105e6c3821471a85@192.168.1.84:5060 [Aug 10 15:46:45] VERBOSE[17897] chan_sip.c: Really destroying SIP dialog '354b26f06bb189d8105e6c3821471a85@192.168.1.84:5060' Method: OPTIONS [Aug 10 15:46:49] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:46:49] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:46:54] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:46:54] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:46:59] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:46:59] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:47:04] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:47:04] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:47:09] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:47:09] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:47:14] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:47:14] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:47:17] DEBUG[17897] chan_sip.c: Auto destroying SIP dialog '50250e9b9684-8d3g9c7o4hy2' [Aug 10 15:47:17] DEBUG[17897] chan_sip.c: Destroying SIP dialog 50250e9b9684-8d3g9c7o4hy2 [Aug 10 15:47:17] VERBOSE[17897] chan_sip.c: Really destroying SIP dialog '50250e9b9684-8d3g9c7o4hy2' Method: REGISTER [Aug 10 15:47:19] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:47:19] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:47:24] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:47:24] DEBUG[17897] chan_sip.c: Allocating new SIP dialog for 6e7782187cd46b6702d43d311091ff1a@127.0.0.1:5060 - OPTIONS (No RTP) [Aug 10 15:47:24] DEBUG[17897] acl.c: For destination '192.168.2.210', our source address is '192.168.1.84'. [Aug 10 15:47:24] DEBUG[17897] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.1.84:5060 [Aug 10 15:47:24] DEBUG[17897] chan_sip.c: Initializing initreq for method OPTIONS - callid 7b72df69205888b779fcd4257b33589d@192.168.1.84:5060 [Aug 10 15:47:24] DEBUG[17897] chan_sip.c: Header 0 [ 43]: OPTIONS sip:2210@192.168.2.210:2048 SIP/2.0 [Aug 10 15:47:24] DEBUG[17897] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK1ce56b60;rport [Aug 10 15:47:24] DEBUG[17897] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 10 15:47:24] DEBUG[17897] chan_sip.c: Header 3 [ 59]: From: "asterisk" ;tag=as496ac28a [Aug 10 15:47:24] DEBUG[17897] chan_sip.c: Header 4 [ 33]: To: [Aug 10 15:47:24] DEBUG[17897] chan_sip.c: Header 5 [ 41]: Contact: [Aug 10 15:47:24] DEBUG[17897] chan_sip.c: Header 6 [ 59]: Call-ID: 7b72df69205888b779fcd4257b33589d@192.168.1.84:5060 [Aug 10 15:47:24] DEBUG[17897] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:47:24] DEBUG[17897] chan_sip.c: Header 8 [ 33]: User-Agent: Asterisk PBX 1.8.15.0 [Aug 10 15:47:24] DEBUG[17897] chan_sip.c: Header 9 [ 35]: Date: Fri, 10 Aug 2012 13:47:24 GMT [Aug 10 15:47:24] DEBUG[17897] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 10 15:47:24] DEBUG[17897] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 10 15:47:24] VERBOSE[17897] chan_sip.c: Reliably Transmitting (NAT) to 192.168.2.210:2048: OPTIONS sip:2210@192.168.2.210:2048 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK1ce56b60;rport Max-Forwards: 70 From: "asterisk" ;tag=as496ac28a To: Contact: Call-ID: 7b72df69205888b779fcd4257b33589d@192.168.1.84:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.15.0 Date: Fri, 10 Aug 2012 13:47:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Aug 10 15:47:24] DEBUG[17897] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #310 [Aug 10 15:47:24] DEBUG[17897] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.2.210:2048 [Aug 10 15:47:24] VERBOSE[17897] chan_sip.c: <--- SIP read from UDP:192.168.2.210:2048 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK1ce56b60;rport=5060 From: "asterisk" ;tag=as496ac28a To: Call-ID: 7b72df69205888b779fcd4257b33589d@192.168.1.84:5060 CSeq: 102 OPTIONS Contact: ;reg-id=1 User-Agent: snom300/8.4.32 Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: 100rel, replaces, from-change Content-Length: 0 <-------------> [Aug 10 15:47:24] DEBUG[17897] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 10 15:47:24] DEBUG[17897] chan_sip.c: Header 1 [ 68]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK1ce56b60;rport=5060 [Aug 10 15:47:24] DEBUG[17897] chan_sip.c: Header 2 [ 59]: From: "asterisk" ;tag=as496ac28a [Aug 10 15:47:24] DEBUG[17897] chan_sip.c: Header 3 [ 33]: To: [Aug 10 15:47:24] DEBUG[17897] chan_sip.c: Header 4 [ 59]: Call-ID: 7b72df69205888b779fcd4257b33589d@192.168.1.84:5060 [Aug 10 15:47:24] DEBUG[17897] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:47:24] DEBUG[17897] chan_sip.c: Header 6 [ 47]: Contact: ;reg-id=1 [Aug 10 15:47:24] DEBUG[17897] chan_sip.c: Header 7 [ 26]: User-Agent: snom300/8.4.32 [Aug 10 15:47:24] DEBUG[17897] chan_sip.c: Header 8 [ 19]: Accept-Language: en [Aug 10 15:47:24] DEBUG[17897] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [Aug 10 15:47:24] DEBUG[17897] chan_sip.c: Header 10 [ 96]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE [Aug 10 15:47:24] DEBUG[17897] chan_sip.c: Header 11 [ 42]: Allow-Events: talk, hold, refer, call-info [Aug 10 15:47:24] DEBUG[17897] chan_sip.c: Header 12 [ 40]: Supported: 100rel, replaces, from-change [Aug 10 15:47:24] DEBUG[17897] chan_sip.c: Header 13 [ 17]: Content-Length: 0 [Aug 10 15:47:24] VERBOSE[17897] chan_sip.c: --- (14 headers 0 lines) --- [Aug 10 15:47:24] DEBUG[17897] chan_sip.c: = Looking for Call ID: 7b72df69205888b779fcd4257b33589d@192.168.1.84:5060 (Checking To) --From tag as496ac28a --To-tag [Aug 10 15:47:24] DEBUG[17897] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #310 [Aug 10 15:47:24] DEBUG[17897] chan_sip.c: Stopping retransmission on '7b72df69205888b779fcd4257b33589d@192.168.1.84:5060' of Request 102: Match Found [Aug 10 15:47:24] DEBUG[17897] chan_sip.c: Destroying SIP dialog 7b72df69205888b779fcd4257b33589d@192.168.1.84:5060 [Aug 10 15:47:24] VERBOSE[17897] chan_sip.c: Really destroying SIP dialog '7b72df69205888b779fcd4257b33589d@192.168.1.84:5060' Method: OPTIONS [Aug 10 15:47:24] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:47:24] DEBUG[17897] chan_sip.c: Allocating new SIP dialog for 1731b3146f71048d334229774c5f3e38@127.0.0.1:5060 - OPTIONS (No RTP) [Aug 10 15:47:24] DEBUG[17897] acl.c: For destination '192.168.1.102', our source address is '192.168.1.84'. [Aug 10 15:47:24] DEBUG[17897] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.1.84:5060 [Aug 10 15:47:24] DEBUG[17897] chan_sip.c: Initializing initreq for method OPTIONS - callid 0a2c01f512d900031f88465e16da7688@192.168.1.84:5060 [Aug 10 15:47:24] DEBUG[17897] chan_sip.c: Header 0 [ 57]: OPTIONS sip:2212@192.168.1.102:2048;line=z1zv7jsj SIP/2.0 [Aug 10 15:47:24] DEBUG[17897] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK13247644;rport [Aug 10 15:47:24] DEBUG[17897] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 10 15:47:24] DEBUG[17897] chan_sip.c: Header 3 [ 59]: From: "asterisk" ;tag=as5f5f1dec [Aug 10 15:47:24] DEBUG[17897] chan_sip.c: Header 4 [ 47]: To: [Aug 10 15:47:24] DEBUG[17897] chan_sip.c: Header 5 [ 41]: Contact: [Aug 10 15:47:24] DEBUG[17897] chan_sip.c: Header 6 [ 59]: Call-ID: 0a2c01f512d900031f88465e16da7688@192.168.1.84:5060 [Aug 10 15:47:24] DEBUG[17897] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:47:24] DEBUG[17897] chan_sip.c: Header 8 [ 33]: User-Agent: Asterisk PBX 1.8.15.0 [Aug 10 15:47:24] DEBUG[17897] chan_sip.c: Header 9 [ 35]: Date: Fri, 10 Aug 2012 13:47:24 GMT [Aug 10 15:47:24] DEBUG[17897] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 10 15:47:24] DEBUG[17897] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 10 15:47:24] VERBOSE[17897] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.102:2048: OPTIONS sip:2212@192.168.1.102:2048;line=z1zv7jsj SIP/2.0 Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK13247644;rport Max-Forwards: 70 From: "asterisk" ;tag=as5f5f1dec To: Contact: Call-ID: 0a2c01f512d900031f88465e16da7688@192.168.1.84:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.15.0 Date: Fri, 10 Aug 2012 13:47:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Aug 10 15:47:24] DEBUG[17897] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #313 [Aug 10 15:47:24] DEBUG[17897] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.1.102:2048 [Aug 10 15:47:24] VERBOSE[17897] chan_sip.c: <--- SIP read from UDP:192.168.1.102:2048 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK13247644;rport=5060 From: "asterisk" ;tag=as5f5f1dec To: Call-ID: 0a2c01f512d900031f88465e16da7688@192.168.1.84:5060 CSeq: 102 OPTIONS Contact: ;reg-id=1 User-Agent: snom300/8.4.32 Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Content-Length: 0 <-------------> [Aug 10 15:47:24] DEBUG[17897] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 10 15:47:24] DEBUG[17897] chan_sip.c: Header 1 [ 68]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK13247644;rport=5060 [Aug 10 15:47:24] DEBUG[17897] chan_sip.c: Header 2 [ 59]: From: "asterisk" ;tag=as5f5f1dec [Aug 10 15:47:24] DEBUG[17897] chan_sip.c: Header 3 [ 47]: To: [Aug 10 15:47:24] DEBUG[17897] chan_sip.c: Header 4 [ 59]: Call-ID: 0a2c01f512d900031f88465e16da7688@192.168.1.84:5060 [Aug 10 15:47:24] DEBUG[17897] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:47:24] DEBUG[17897] chan_sip.c: Header 6 [ 61]: Contact: ;reg-id=1 [Aug 10 15:47:24] DEBUG[17897] chan_sip.c: Header 7 [ 26]: User-Agent: snom300/8.4.32 [Aug 10 15:47:24] DEBUG[17897] chan_sip.c: Header 8 [ 19]: Accept-Language: en [Aug 10 15:47:24] DEBUG[17897] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [Aug 10 15:47:24] DEBUG[17897] chan_sip.c: Header 10 [ 96]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE [Aug 10 15:47:24] DEBUG[17897] chan_sip.c: Header 11 [ 42]: Allow-Events: talk, hold, refer, call-info [Aug 10 15:47:24] DEBUG[17897] chan_sip.c: Header 12 [ 47]: Supported: timer, 100rel, replaces, from-change [Aug 10 15:47:24] DEBUG[17897] chan_sip.c: Header 13 [ 17]: Content-Length: 0 [Aug 10 15:47:24] VERBOSE[17897] chan_sip.c: --- (14 headers 0 lines) --- [Aug 10 15:47:24] DEBUG[17897] chan_sip.c: = Looking for Call ID: 0a2c01f512d900031f88465e16da7688@192.168.1.84:5060 (Checking To) --From tag as5f5f1dec --To-tag [Aug 10 15:47:24] DEBUG[17897] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #313 [Aug 10 15:47:24] DEBUG[17897] chan_sip.c: Stopping retransmission on '0a2c01f512d900031f88465e16da7688@192.168.1.84:5060' of Request 102: Match Found [Aug 10 15:47:24] DEBUG[17897] chan_sip.c: Destroying SIP dialog 0a2c01f512d900031f88465e16da7688@192.168.1.84:5060 [Aug 10 15:47:24] VERBOSE[17897] chan_sip.c: Really destroying SIP dialog '0a2c01f512d900031f88465e16da7688@192.168.1.84:5060' Method: OPTIONS [Aug 10 15:47:29] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:47:29] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:47:34] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:47:34] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:47:39] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:47:39] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:47:44] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:47:44] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: Allocating new SIP dialog for 603a1c4534c0eef5231dc18a1afd0bdf@127.0.0.1:5060 - OPTIONS (No RTP) [Aug 10 15:47:45] DEBUG[17897] acl.c: For destination '192.168.1.106', our source address is '192.168.1.84'. [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.1.84:5060 [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: Initializing initreq for method OPTIONS - callid 147fe7fe4598f04d3a99931121e43722@192.168.1.84:5060 [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: Header 0 [ 57]: OPTIONS sip:2219@192.168.1.106:2048;line=7g8ggsvj SIP/2.0 [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK76f1dbdc;rport [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: Header 3 [ 59]: From: "asterisk" ;tag=as3976961a [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: Header 4 [ 47]: To: [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: Header 5 [ 41]: Contact: [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: Header 6 [ 59]: Call-ID: 147fe7fe4598f04d3a99931121e43722@192.168.1.84:5060 [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: Header 8 [ 33]: User-Agent: Asterisk PBX 1.8.15.0 [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: Header 9 [ 35]: Date: Fri, 10 Aug 2012 13:47:45 GMT [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 10 15:47:45] VERBOSE[17897] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.106:2048: OPTIONS sip:2219@192.168.1.106:2048;line=7g8ggsvj SIP/2.0 Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK76f1dbdc;rport Max-Forwards: 70 From: "asterisk" ;tag=as3976961a To: Contact: Call-ID: 147fe7fe4598f04d3a99931121e43722@192.168.1.84:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.15.0 Date: Fri, 10 Aug 2012 13:47:45 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #316 [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.1.106:2048 [Aug 10 15:47:45] VERBOSE[17897] chan_sip.c: <--- SIP read from UDP:192.168.1.106:2048 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK76f1dbdc;rport=5060 From: "asterisk" ;tag=as3976961a To: ;tag=soyh897wp3 Call-ID: 147fe7fe4598f04d3a99931121e43722@192.168.1.84:5060 CSeq: 102 OPTIONS Contact: ;reg-id=1 User-Agent: snom300/8.7.3.10 Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: 100rel, replaces, from-change Content-Length: 0 <-------------> [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: Header 1 [ 68]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK76f1dbdc;rport=5060 [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: Header 2 [ 59]: From: "asterisk" ;tag=as3976961a [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: Header 3 [ 62]: To: ;tag=soyh897wp3 [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: Header 4 [ 59]: Call-ID: 147fe7fe4598f04d3a99931121e43722@192.168.1.84:5060 [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: Header 6 [ 61]: Contact: ;reg-id=1 [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: Header 7 [ 28]: User-Agent: snom300/8.7.3.10 [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: Header 8 [ 19]: Accept-Language: en [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: Header 10 [ 96]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: Header 11 [ 42]: Allow-Events: talk, hold, refer, call-info [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: Header 12 [ 40]: Supported: 100rel, replaces, from-change [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: Header 13 [ 17]: Content-Length: 0 [Aug 10 15:47:45] VERBOSE[17897] chan_sip.c: --- (14 headers 0 lines) --- [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: = Looking for Call ID: 147fe7fe4598f04d3a99931121e43722@192.168.1.84:5060 (Checking To) --From tag as3976961a --To-tag soyh897wp3 [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #316 [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: Stopping retransmission on '147fe7fe4598f04d3a99931121e43722@192.168.1.84:5060' of Request 102: Match Found [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: Destroying SIP dialog 147fe7fe4598f04d3a99931121e43722@192.168.1.84:5060 [Aug 10 15:47:45] VERBOSE[17897] chan_sip.c: Really destroying SIP dialog '147fe7fe4598f04d3a99931121e43722@192.168.1.84:5060' Method: OPTIONS [Aug 10 15:47:45] VERBOSE[17897] chan_sip.c: <--- SIP read from UDP:192.168.1.106:2048 ---> REGISTER sip:192.168.1.84 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.106:2048;branch=z9hG4bK-l705h0ky5cxt;rport From: "2219" ;tag=pqb59r72wl To: "2219" Call-ID: 50250e9b9684-8d3g9c7o4hy2 CSeq: 27 REGISTER Max-Forwards: 70 Contact: ;reg-id=1;q=1.0;+sip.instance="";audio;mobility="fixed";duplex="full";description="snom300";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO" User-Agent: snom300/8.7.3.10 Allow-Events: dialog X-Real-IP: 192.168.1.106 Supported: path, gruu Content-Length: 0 <-------------> [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: Header 0 [ 33]: REGISTER sip:192.168.1.84 SIP/2.0 [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: Header 1 [ 69]: Via: SIP/2.0/UDP 192.168.1.106:2048;branch=z9hG4bK-l705h0ky5cxt;rport [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: Header 2 [ 51]: From: "2219" ;tag=pqb59r72wl [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: Header 3 [ 34]: To: "2219" [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: Header 4 [ 34]: Call-ID: 50250e9b9684-8d3g9c7o4hy2 [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: Header 5 [ 17]: CSeq: 27 REGISTER [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: Header 7 [306]: Contact: ;reg-id=1;q=1.0;+sip.instance="";audio;mobility="fixed";duplex="full";description="snom300";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO" [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: Header 8 [ 28]: User-Agent: snom300/8.7.3.10 [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: Header 9 [ 20]: Allow-Events: dialog [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: Header 10 [ 24]: X-Real-IP: 192.168.1.106 [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: Header 11 [ 21]: Supported: path, gruu [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Aug 10 15:47:45] VERBOSE[17897] chan_sip.c: --- (13 headers 0 lines) --- [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: = Looking for Call ID: 50250e9b9684-8d3g9c7o4hy2 (Checking From) --From tag pqb59r72wl --To-tag [Aug 10 15:47:45] DEBUG[17897] acl.c: For destination '192.168.1.106', our source address is '192.168.1.84'. [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.1.84:5060 [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: Allocating new SIP dialog for 50250e9b9684-8d3g9c7o4hy2 - REGISTER (No RTP) [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: Initializing initreq for method REGISTER - callid 50250e9b9684-8d3g9c7o4hy2 [Aug 10 15:47:45] DEBUG[17897] netsock2.c: Splitting '192.168.1.106:2048' into... [Aug 10 15:47:45] DEBUG[17897] netsock2.c: ...host '192.168.1.106' and port '2048'. [Aug 10 15:47:45] VERBOSE[17897] chan_sip.c: Sending to 192.168.1.106:2048 (NAT) [Aug 10 15:47:45] DEBUG[17897] netsock2.c: Splitting '192.168.1.84' into... [Aug 10 15:47:45] DEBUG[17897] netsock2.c: ...host '192.168.1.84' and port ''. [Aug 10 15:47:45] VERBOSE[17897] chan_sip.c: <--- Transmitting (NAT) to 192.168.1.106:2048 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.106:2048;branch=z9hG4bK-l705h0ky5cxt;received=192.168.1.106;rport=2048 From: "2219" ;tag=pqb59r72wl To: "2219" ;tag=as16412d35 Call-ID: 50250e9b9684-8d3g9c7o4hy2 CSeq: 27 REGISTER Server: Asterisk PBX 1.8.15.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3fdfafad" Content-Length: 0 <------------> [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 192.168.1.106:2048 [Aug 10 15:47:45] VERBOSE[17897] chan_sip.c: Scheduling destruction of SIP dialog '50250e9b9684-8d3g9c7o4hy2' in 32000 ms (Method: REGISTER) [Aug 10 15:47:45] VERBOSE[17897] chan_sip.c: <--- SIP read from UDP:192.168.1.106:2048 ---> REGISTER sip:192.168.1.84 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.106:2048;branch=z9hG4bK-bc7g9tntipum;rport From: "2219" ;tag=pqb59r72wl To: "2219" Call-ID: 50250e9b9684-8d3g9c7o4hy2 CSeq: 28 REGISTER Max-Forwards: 70 Contact: ;reg-id=1;q=1.0;+sip.instance="";audio;mobility="fixed";duplex="full";description="snom300";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO" User-Agent: snom300/8.7.3.10 Allow-Events: dialog X-Real-IP: 192.168.1.106 Supported: path, gruu Authorization: Digest username="2219",realm="asterisk",nonce="3fdfafad",uri="sip:192.168.1.84",response="34a5be2fb38acaa6e78f920d168e605b",algorithm=MD5 Content-Length: 0 <-------------> [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: Header 0 [ 33]: REGISTER sip:192.168.1.84 SIP/2.0 [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: Header 1 [ 69]: Via: SIP/2.0/UDP 192.168.1.106:2048;branch=z9hG4bK-bc7g9tntipum;rport [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: Header 2 [ 51]: From: "2219" ;tag=pqb59r72wl [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: Header 3 [ 34]: To: "2219" [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: Header 4 [ 34]: Call-ID: 50250e9b9684-8d3g9c7o4hy2 [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: Header 5 [ 17]: CSeq: 28 REGISTER [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: Header 7 [306]: Contact: ;reg-id=1;q=1.0;+sip.instance="";audio;mobility="fixed";duplex="full";description="snom300";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO" [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: Header 8 [ 28]: User-Agent: snom300/8.7.3.10 [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: Header 9 [ 20]: Allow-Events: dialog [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: Header 10 [ 24]: X-Real-IP: 192.168.1.106 [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: Header 11 [ 21]: Supported: path, gruu [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: Header 12 [152]: Authorization: Digest username="2219",realm="asterisk",nonce="3fdfafad",uri="sip:192.168.1.84",response="34a5be2fb38acaa6e78f920d168e605b",algorithm=MD5 [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: Header 13 [ 17]: Content-Length: 0 [Aug 10 15:47:45] VERBOSE[17897] chan_sip.c: --- (14 headers 0 lines) --- [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: = Looking for Call ID: 50250e9b9684-8d3g9c7o4hy2 (Checking From) --From tag pqb59r72wl --To-tag [Aug 10 15:47:45] DEBUG[17897] netsock2.c: Splitting '192.168.1.84' into... [Aug 10 15:47:45] DEBUG[17897] netsock2.c: ...host '192.168.1.84' and port ''. [Aug 10 15:47:45] DEBUG[17897] netsock2.c: Splitting '192.168.1.84' into... [Aug 10 15:47:45] DEBUG[17897] netsock2.c: ...host '192.168.1.84' and port ''. [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: Initializing initreq for method REGISTER - callid 50250e9b9684-8d3g9c7o4hy2 [Aug 10 15:47:45] DEBUG[17897] netsock2.c: Splitting '192.168.1.106:2048' into... [Aug 10 15:47:45] DEBUG[17897] netsock2.c: ...host '192.168.1.106' and port '2048'. [Aug 10 15:47:45] VERBOSE[17897] chan_sip.c: Sending to 192.168.1.106:2048 (NAT) [Aug 10 15:47:45] DEBUG[17897] netsock2.c: Splitting '192.168.1.84' into... [Aug 10 15:47:45] DEBUG[17897] netsock2.c: ...host '192.168.1.84' and port ''. [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: Store REGISTER's src-IP:port for call routing. [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: Allocating new SIP dialog for 4d21c0f613d9fc9e55ade7cc5d0ade90@127.0.0.1:5060 - OPTIONS (No RTP) [Aug 10 15:47:45] DEBUG[17897] acl.c: For destination '192.168.1.106', our source address is '192.168.1.84'. [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.1.84:5060 [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: Initializing initreq for method OPTIONS - callid 6e78b4e21d8d02990b3ab4bd31f25e7a@192.168.1.84:5060 [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: Header 0 [ 57]: OPTIONS sip:2219@192.168.1.106:2048;line=7g8ggsvj SIP/2.0 [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK0456f2cf;rport [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: Header 3 [ 59]: From: "asterisk" ;tag=as3d66ba02 [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: Header 4 [ 47]: To: [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: Header 5 [ 41]: Contact: [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: Header 6 [ 59]: Call-ID: 6e78b4e21d8d02990b3ab4bd31f25e7a@192.168.1.84:5060 [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: Header 8 [ 33]: User-Agent: Asterisk PBX 1.8.15.0 [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: Header 9 [ 35]: Date: Fri, 10 Aug 2012 13:47:45 GMT [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 10 15:47:45] VERBOSE[17897] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.106:2048: OPTIONS sip:2219@192.168.1.106:2048;line=7g8ggsvj SIP/2.0 Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK0456f2cf;rport Max-Forwards: 70 From: "asterisk" ;tag=as3d66ba02 To: Contact: Call-ID: 6e78b4e21d8d02990b3ab4bd31f25e7a@192.168.1.84:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.15.0 Date: Fri, 10 Aug 2012 13:47:45 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #321 [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.1.106:2048 [Aug 10 15:47:45] VERBOSE[17897] chan_sip.c: <--- Transmitting (NAT) to 192.168.1.106:2048 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.106:2048;branch=z9hG4bK-bc7g9tntipum;received=192.168.1.106;rport=2048 From: "2219" ;tag=pqb59r72wl To: "2219" ;tag=as16412d35 Call-ID: 50250e9b9684-8d3g9c7o4hy2 CSeq: 28 REGISTER Server: Asterisk PBX 1.8.15.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Expires: 120 Contact: ;expires=120 Date: Fri, 10 Aug 2012 13:47:45 GMT Content-Length: 0 <------------> [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.1.106:2048 [Aug 10 15:47:45] VERBOSE[17897] chan_sip.c: Scheduling destruction of SIP dialog '50250e9b9684-8d3g9c7o4hy2' in 32000 ms (Method: REGISTER) [Aug 10 15:47:45] DEBUG[17874] devicestate.c: No provider found, checking channel drivers for SIP - 2219 [Aug 10 15:47:45] DEBUG[17874] chan_sip.c: Checking device state for peer 2219 [Aug 10 15:47:45] DEBUG[17874] devicestate.c: Changing state for SIP/2219 - state 1 (Not in use) [Aug 10 15:47:45] DEBUG[17874] devicestate.c: device 'SIP/2219' state '1' [Aug 10 15:47:45] DEBUG[17907] app_queue.c: Device 'SIP/2219' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Aug 10 15:47:45] VERBOSE[17897] chan_sip.c: <--- SIP read from UDP:192.168.1.106:2048 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK0456f2cf;rport=5060 From: "asterisk" ;tag=as3d66ba02 To: ;tag=nb29xfofgw Call-ID: 6e78b4e21d8d02990b3ab4bd31f25e7a@192.168.1.84:5060 CSeq: 102 OPTIONS Contact: ;reg-id=1 User-Agent: snom300/8.7.3.10 Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: 100rel, replaces, from-change Content-Length: 0 <-------------> [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: Header 1 [ 68]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK0456f2cf;rport=5060 [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: Header 2 [ 59]: From: "asterisk" ;tag=as3d66ba02 [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: Header 3 [ 62]: To: ;tag=nb29xfofgw [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: Header 4 [ 59]: Call-ID: 6e78b4e21d8d02990b3ab4bd31f25e7a@192.168.1.84:5060 [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: Header 6 [ 61]: Contact: ;reg-id=1 [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: Header 7 [ 28]: User-Agent: snom300/8.7.3.10 [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: Header 8 [ 19]: Accept-Language: en [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: Header 10 [ 96]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: Header 11 [ 42]: Allow-Events: talk, hold, refer, call-info [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: Header 12 [ 40]: Supported: 100rel, replaces, from-change [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: Header 13 [ 17]: Content-Length: 0 [Aug 10 15:47:45] VERBOSE[17897] chan_sip.c: --- (14 headers 0 lines) --- [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: = Looking for Call ID: 6e78b4e21d8d02990b3ab4bd31f25e7a@192.168.1.84:5060 (Checking To) --From tag as3d66ba02 --To-tag nb29xfofgw [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #321 [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: Stopping retransmission on '6e78b4e21d8d02990b3ab4bd31f25e7a@192.168.1.84:5060' of Request 102: Match Found [Aug 10 15:47:45] DEBUG[17897] chan_sip.c: Destroying SIP dialog 6e78b4e21d8d02990b3ab4bd31f25e7a@192.168.1.84:5060 [Aug 10 15:47:45] VERBOSE[17897] chan_sip.c: Really destroying SIP dialog '6e78b4e21d8d02990b3ab4bd31f25e7a@192.168.1.84:5060' Method: OPTIONS [Aug 10 15:47:49] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:47:49] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:47:54] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:47:54] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:47:59] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:47:59] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:48:04] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:48:04] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:48:09] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:48:09] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:48:14] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:48:14] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:48:17] DEBUG[17897] chan_sip.c: Auto destroying SIP dialog '50250e9b9684-8d3g9c7o4hy2' [Aug 10 15:48:17] DEBUG[17897] chan_sip.c: Destroying SIP dialog 50250e9b9684-8d3g9c7o4hy2 [Aug 10 15:48:17] VERBOSE[17897] chan_sip.c: Really destroying SIP dialog '50250e9b9684-8d3g9c7o4hy2' Method: REGISTER [Aug 10 15:48:19] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:48:19] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:48:24] DEBUG[17897] chan_sip.c: Allocating new SIP dialog for 2c2c38941859ffea5fa05bba7d3fb8e2@127.0.0.1:5060 - OPTIONS (No RTP) [Aug 10 15:48:24] DEBUG[17897] acl.c: For destination '192.168.2.210', our source address is '192.168.1.84'. [Aug 10 15:48:24] DEBUG[17897] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.1.84:5060 [Aug 10 15:48:24] DEBUG[17897] chan_sip.c: Initializing initreq for method OPTIONS - callid 188ddb8b4d8d83e677bf0a473f0b70c9@192.168.1.84:5060 [Aug 10 15:48:24] DEBUG[17897] chan_sip.c: Header 0 [ 43]: OPTIONS sip:2210@192.168.2.210:2048 SIP/2.0 [Aug 10 15:48:24] DEBUG[17897] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK020baf96;rport [Aug 10 15:48:24] DEBUG[17897] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 10 15:48:24] DEBUG[17897] chan_sip.c: Header 3 [ 59]: From: "asterisk" ;tag=as2a02b8c9 [Aug 10 15:48:24] DEBUG[17897] chan_sip.c: Header 4 [ 33]: To: [Aug 10 15:48:24] DEBUG[17897] chan_sip.c: Header 5 [ 41]: Contact: [Aug 10 15:48:24] DEBUG[17897] chan_sip.c: Header 6 [ 59]: Call-ID: 188ddb8b4d8d83e677bf0a473f0b70c9@192.168.1.84:5060 [Aug 10 15:48:24] DEBUG[17897] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:48:24] DEBUG[17897] chan_sip.c: Header 8 [ 33]: User-Agent: Asterisk PBX 1.8.15.0 [Aug 10 15:48:24] DEBUG[17897] chan_sip.c: Header 9 [ 35]: Date: Fri, 10 Aug 2012 13:48:24 GMT [Aug 10 15:48:24] DEBUG[17897] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 10 15:48:24] DEBUG[17897] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 10 15:48:24] VERBOSE[17897] chan_sip.c: Reliably Transmitting (NAT) to 192.168.2.210:2048: OPTIONS sip:2210@192.168.2.210:2048 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK020baf96;rport Max-Forwards: 70 From: "asterisk" ;tag=as2a02b8c9 To: Contact: Call-ID: 188ddb8b4d8d83e677bf0a473f0b70c9@192.168.1.84:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.15.0 Date: Fri, 10 Aug 2012 13:48:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Aug 10 15:48:24] DEBUG[17897] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #325 [Aug 10 15:48:24] DEBUG[17897] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.2.210:2048 [Aug 10 15:48:24] VERBOSE[17897] chan_sip.c: <--- SIP read from UDP:192.168.2.210:2048 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK020baf96;rport=5060 From: "asterisk" ;tag=as2a02b8c9 To: Call-ID: 188ddb8b4d8d83e677bf0a473f0b70c9@192.168.1.84:5060 CSeq: 102 OPTIONS Contact: ;reg-id=1 User-Agent: snom300/8.4.32 Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: 100rel, replaces, from-change Content-Length: 0 <-------------> [Aug 10 15:48:24] DEBUG[17897] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 10 15:48:24] DEBUG[17897] chan_sip.c: Header 1 [ 68]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK020baf96;rport=5060 [Aug 10 15:48:24] DEBUG[17897] chan_sip.c: Header 2 [ 59]: From: "asterisk" ;tag=as2a02b8c9 [Aug 10 15:48:24] DEBUG[17897] chan_sip.c: Header 3 [ 33]: To: [Aug 10 15:48:24] DEBUG[17897] chan_sip.c: Header 4 [ 59]: Call-ID: 188ddb8b4d8d83e677bf0a473f0b70c9@192.168.1.84:5060 [Aug 10 15:48:24] DEBUG[17897] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:48:24] DEBUG[17897] chan_sip.c: Header 6 [ 47]: Contact: ;reg-id=1 [Aug 10 15:48:24] DEBUG[17897] chan_sip.c: Header 7 [ 26]: User-Agent: snom300/8.4.32 [Aug 10 15:48:24] DEBUG[17897] chan_sip.c: Header 8 [ 19]: Accept-Language: en [Aug 10 15:48:24] DEBUG[17897] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [Aug 10 15:48:24] DEBUG[17897] chan_sip.c: Header 10 [ 96]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE [Aug 10 15:48:24] DEBUG[17897] chan_sip.c: Header 11 [ 42]: Allow-Events: talk, hold, refer, call-info [Aug 10 15:48:24] DEBUG[17897] chan_sip.c: Header 12 [ 40]: Supported: 100rel, replaces, from-change [Aug 10 15:48:24] DEBUG[17897] chan_sip.c: Header 13 [ 17]: Content-Length: 0 [Aug 10 15:48:24] VERBOSE[17897] chan_sip.c: --- (14 headers 0 lines) --- [Aug 10 15:48:24] DEBUG[17897] chan_sip.c: = Looking for Call ID: 188ddb8b4d8d83e677bf0a473f0b70c9@192.168.1.84:5060 (Checking To) --From tag as2a02b8c9 --To-tag [Aug 10 15:48:24] DEBUG[17897] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #325 [Aug 10 15:48:24] DEBUG[17897] chan_sip.c: Stopping retransmission on '188ddb8b4d8d83e677bf0a473f0b70c9@192.168.1.84:5060' of Request 102: Match Found [Aug 10 15:48:24] DEBUG[17897] chan_sip.c: Destroying SIP dialog 188ddb8b4d8d83e677bf0a473f0b70c9@192.168.1.84:5060 [Aug 10 15:48:24] VERBOSE[17897] chan_sip.c: Really destroying SIP dialog '188ddb8b4d8d83e677bf0a473f0b70c9@192.168.1.84:5060' Method: OPTIONS [Aug 10 15:48:24] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:48:24] DEBUG[17897] chan_sip.c: Allocating new SIP dialog for 29cb59d214934b2c55de89e55b1d8a84@127.0.0.1:5060 - OPTIONS (No RTP) [Aug 10 15:48:24] DEBUG[17897] acl.c: For destination '192.168.1.102', our source address is '192.168.1.84'. [Aug 10 15:48:24] DEBUG[17897] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.1.84:5060 [Aug 10 15:48:24] DEBUG[17897] chan_sip.c: Initializing initreq for method OPTIONS - callid 1bc4084c7edbea65437a29e03869f255@192.168.1.84:5060 [Aug 10 15:48:24] DEBUG[17897] chan_sip.c: Header 0 [ 57]: OPTIONS sip:2212@192.168.1.102:2048;line=z1zv7jsj SIP/2.0 [Aug 10 15:48:24] DEBUG[17897] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK6032cf10;rport [Aug 10 15:48:24] DEBUG[17897] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 10 15:48:24] DEBUG[17897] chan_sip.c: Header 3 [ 59]: From: "asterisk" ;tag=as2a00515b [Aug 10 15:48:24] DEBUG[17897] chan_sip.c: Header 4 [ 47]: To: [Aug 10 15:48:24] DEBUG[17897] chan_sip.c: Header 5 [ 41]: Contact: [Aug 10 15:48:24] DEBUG[17897] chan_sip.c: Header 6 [ 59]: Call-ID: 1bc4084c7edbea65437a29e03869f255@192.168.1.84:5060 [Aug 10 15:48:24] DEBUG[17897] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:48:24] DEBUG[17897] chan_sip.c: Header 8 [ 33]: User-Agent: Asterisk PBX 1.8.15.0 [Aug 10 15:48:24] DEBUG[17897] chan_sip.c: Header 9 [ 35]: Date: Fri, 10 Aug 2012 13:48:24 GMT [Aug 10 15:48:24] DEBUG[17897] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 10 15:48:24] DEBUG[17897] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 10 15:48:24] VERBOSE[17897] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.102:2048: OPTIONS sip:2212@192.168.1.102:2048;line=z1zv7jsj SIP/2.0 Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK6032cf10;rport Max-Forwards: 70 From: "asterisk" ;tag=as2a00515b To: Contact: Call-ID: 1bc4084c7edbea65437a29e03869f255@192.168.1.84:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.15.0 Date: Fri, 10 Aug 2012 13:48:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Aug 10 15:48:24] DEBUG[17897] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #328 [Aug 10 15:48:24] DEBUG[17897] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.1.102:2048 [Aug 10 15:48:24] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:48:24] VERBOSE[17897] chan_sip.c: <--- SIP read from UDP:192.168.1.102:2048 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK6032cf10;rport=5060 From: "asterisk" ;tag=as2a00515b To: Call-ID: 1bc4084c7edbea65437a29e03869f255@192.168.1.84:5060 CSeq: 102 OPTIONS Contact: ;reg-id=1 User-Agent: snom300/8.4.32 Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Content-Length: 0 <-------------> [Aug 10 15:48:24] DEBUG[17897] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 10 15:48:24] DEBUG[17897] chan_sip.c: Header 1 [ 68]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK6032cf10;rport=5060 [Aug 10 15:48:24] DEBUG[17897] chan_sip.c: Header 2 [ 59]: From: "asterisk" ;tag=as2a00515b [Aug 10 15:48:24] DEBUG[17897] chan_sip.c: Header 3 [ 47]: To: [Aug 10 15:48:24] DEBUG[17897] chan_sip.c: Header 4 [ 59]: Call-ID: 1bc4084c7edbea65437a29e03869f255@192.168.1.84:5060 [Aug 10 15:48:24] DEBUG[17897] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:48:24] DEBUG[17897] chan_sip.c: Header 6 [ 61]: Contact: ;reg-id=1 [Aug 10 15:48:24] DEBUG[17897] chan_sip.c: Header 7 [ 26]: User-Agent: snom300/8.4.32 [Aug 10 15:48:24] DEBUG[17897] chan_sip.c: Header 8 [ 19]: Accept-Language: en [Aug 10 15:48:24] DEBUG[17897] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [Aug 10 15:48:24] DEBUG[17897] chan_sip.c: Header 10 [ 96]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE [Aug 10 15:48:24] DEBUG[17897] chan_sip.c: Header 11 [ 42]: Allow-Events: talk, hold, refer, call-info [Aug 10 15:48:24] DEBUG[17897] chan_sip.c: Header 12 [ 47]: Supported: timer, 100rel, replaces, from-change [Aug 10 15:48:24] DEBUG[17897] chan_sip.c: Header 13 [ 17]: Content-Length: 0 [Aug 10 15:48:24] VERBOSE[17897] chan_sip.c: --- (14 headers 0 lines) --- [Aug 10 15:48:24] DEBUG[17897] chan_sip.c: = Looking for Call ID: 1bc4084c7edbea65437a29e03869f255@192.168.1.84:5060 (Checking To) --From tag as2a00515b --To-tag [Aug 10 15:48:24] DEBUG[17897] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #328 [Aug 10 15:48:24] DEBUG[17897] chan_sip.c: Stopping retransmission on '1bc4084c7edbea65437a29e03869f255@192.168.1.84:5060' of Request 102: Match Found [Aug 10 15:48:24] DEBUG[17897] chan_sip.c: Destroying SIP dialog 1bc4084c7edbea65437a29e03869f255@192.168.1.84:5060 [Aug 10 15:48:24] VERBOSE[17897] chan_sip.c: Really destroying SIP dialog '1bc4084c7edbea65437a29e03869f255@192.168.1.84:5060' Method: OPTIONS [Aug 10 15:48:29] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:48:29] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:48:34] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:48:34] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:48:39] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:48:39] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:48:44] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:48:44] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: Allocating new SIP dialog for 47fc50d34513a92863a3b0eb6ee8a13e@127.0.0.1:5060 - OPTIONS (No RTP) [Aug 10 15:48:45] DEBUG[17897] acl.c: For destination '192.168.1.106', our source address is '192.168.1.84'. [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.1.84:5060 [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: Initializing initreq for method OPTIONS - callid 0dfb49db24723ffc588aab2672c34b2d@192.168.1.84:5060 [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: Header 0 [ 57]: OPTIONS sip:2219@192.168.1.106:2048;line=7g8ggsvj SIP/2.0 [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK48e6573e;rport [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: Header 3 [ 59]: From: "asterisk" ;tag=as57e46499 [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: Header 4 [ 47]: To: [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: Header 5 [ 41]: Contact: [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: Header 6 [ 59]: Call-ID: 0dfb49db24723ffc588aab2672c34b2d@192.168.1.84:5060 [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: Header 8 [ 33]: User-Agent: Asterisk PBX 1.8.15.0 [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: Header 9 [ 35]: Date: Fri, 10 Aug 2012 13:48:45 GMT [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 10 15:48:45] VERBOSE[17897] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.106:2048: OPTIONS sip:2219@192.168.1.106:2048;line=7g8ggsvj SIP/2.0 Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK48e6573e;rport Max-Forwards: 70 From: "asterisk" ;tag=as57e46499 To: Contact: Call-ID: 0dfb49db24723ffc588aab2672c34b2d@192.168.1.84:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.15.0 Date: Fri, 10 Aug 2012 13:48:45 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #331 [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.1.106:2048 [Aug 10 15:48:45] VERBOSE[17897] chan_sip.c: <--- SIP read from UDP:192.168.1.106:2048 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK48e6573e;rport=5060 From: "asterisk" ;tag=as57e46499 To: ;tag=mb53cshqwf Call-ID: 0dfb49db24723ffc588aab2672c34b2d@192.168.1.84:5060 CSeq: 102 OPTIONS Contact: ;reg-id=1 User-Agent: snom300/8.7.3.10 Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: 100rel, replaces, from-change Content-Length: 0 <-------------> [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: Header 1 [ 68]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK48e6573e;rport=5060 [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: Header 2 [ 59]: From: "asterisk" ;tag=as57e46499 [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: Header 3 [ 62]: To: ;tag=mb53cshqwf [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: Header 4 [ 59]: Call-ID: 0dfb49db24723ffc588aab2672c34b2d@192.168.1.84:5060 [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: Header 6 [ 61]: Contact: ;reg-id=1 [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: Header 7 [ 28]: User-Agent: snom300/8.7.3.10 [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: Header 8 [ 19]: Accept-Language: en [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: Header 10 [ 96]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: Header 11 [ 42]: Allow-Events: talk, hold, refer, call-info [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: Header 12 [ 40]: Supported: 100rel, replaces, from-change [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: Header 13 [ 17]: Content-Length: 0 [Aug 10 15:48:45] VERBOSE[17897] chan_sip.c: --- (14 headers 0 lines) --- [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: = Looking for Call ID: 0dfb49db24723ffc588aab2672c34b2d@192.168.1.84:5060 (Checking To) --From tag as57e46499 --To-tag mb53cshqwf [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #331 [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: Stopping retransmission on '0dfb49db24723ffc588aab2672c34b2d@192.168.1.84:5060' of Request 102: Match Found [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: Destroying SIP dialog 0dfb49db24723ffc588aab2672c34b2d@192.168.1.84:5060 [Aug 10 15:48:45] VERBOSE[17897] chan_sip.c: Really destroying SIP dialog '0dfb49db24723ffc588aab2672c34b2d@192.168.1.84:5060' Method: OPTIONS [Aug 10 15:48:45] VERBOSE[17897] chan_sip.c: <--- SIP read from UDP:192.168.1.106:2048 ---> REGISTER sip:192.168.1.84 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.106:2048;branch=z9hG4bK-jy1ajrddppxz;rport From: "2219" ;tag=klq3amiyhf To: "2219" Call-ID: 50250e9b9684-8d3g9c7o4hy2 CSeq: 29 REGISTER Max-Forwards: 70 Contact: ;reg-id=1;q=1.0;+sip.instance="";audio;mobility="fixed";duplex="full";description="snom300";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO" User-Agent: snom300/8.7.3.10 Allow-Events: dialog X-Real-IP: 192.168.1.106 Supported: path, gruu Content-Length: 0 <-------------> [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: Header 0 [ 33]: REGISTER sip:192.168.1.84 SIP/2.0 [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: Header 1 [ 69]: Via: SIP/2.0/UDP 192.168.1.106:2048;branch=z9hG4bK-jy1ajrddppxz;rport [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: Header 2 [ 51]: From: "2219" ;tag=klq3amiyhf [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: Header 3 [ 34]: To: "2219" [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: Header 4 [ 34]: Call-ID: 50250e9b9684-8d3g9c7o4hy2 [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: Header 5 [ 17]: CSeq: 29 REGISTER [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: Header 7 [306]: Contact: ;reg-id=1;q=1.0;+sip.instance="";audio;mobility="fixed";duplex="full";description="snom300";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO" [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: Header 8 [ 28]: User-Agent: snom300/8.7.3.10 [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: Header 9 [ 20]: Allow-Events: dialog [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: Header 10 [ 24]: X-Real-IP: 192.168.1.106 [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: Header 11 [ 21]: Supported: path, gruu [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Aug 10 15:48:45] VERBOSE[17897] chan_sip.c: --- (13 headers 0 lines) --- [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: = Looking for Call ID: 50250e9b9684-8d3g9c7o4hy2 (Checking From) --From tag klq3amiyhf --To-tag [Aug 10 15:48:45] DEBUG[17897] acl.c: For destination '192.168.1.106', our source address is '192.168.1.84'. [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.1.84:5060 [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: Allocating new SIP dialog for 50250e9b9684-8d3g9c7o4hy2 - REGISTER (No RTP) [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: Initializing initreq for method REGISTER - callid 50250e9b9684-8d3g9c7o4hy2 [Aug 10 15:48:45] DEBUG[17897] netsock2.c: Splitting '192.168.1.106:2048' into... [Aug 10 15:48:45] DEBUG[17897] netsock2.c: ...host '192.168.1.106' and port '2048'. [Aug 10 15:48:45] VERBOSE[17897] chan_sip.c: Sending to 192.168.1.106:2048 (NAT) [Aug 10 15:48:45] DEBUG[17897] netsock2.c: Splitting '192.168.1.84' into... [Aug 10 15:48:45] DEBUG[17897] netsock2.c: ...host '192.168.1.84' and port ''. [Aug 10 15:48:45] VERBOSE[17897] chan_sip.c: <--- Transmitting (NAT) to 192.168.1.106:2048 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.106:2048;branch=z9hG4bK-jy1ajrddppxz;received=192.168.1.106;rport=2048 From: "2219" ;tag=klq3amiyhf To: "2219" ;tag=as0cf23243 Call-ID: 50250e9b9684-8d3g9c7o4hy2 CSeq: 29 REGISTER Server: Asterisk PBX 1.8.15.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="60b089fe" Content-Length: 0 <------------> [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 192.168.1.106:2048 [Aug 10 15:48:45] VERBOSE[17897] chan_sip.c: Scheduling destruction of SIP dialog '50250e9b9684-8d3g9c7o4hy2' in 32000 ms (Method: REGISTER) [Aug 10 15:48:45] VERBOSE[17897] chan_sip.c: <--- SIP read from UDP:192.168.1.106:2048 ---> REGISTER sip:192.168.1.84 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.106:2048;branch=z9hG4bK-byqk1t2m4mx1;rport From: "2219" ;tag=klq3amiyhf To: "2219" Call-ID: 50250e9b9684-8d3g9c7o4hy2 CSeq: 30 REGISTER Max-Forwards: 70 Contact: ;reg-id=1;q=1.0;+sip.instance="";audio;mobility="fixed";duplex="full";description="snom300";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO" User-Agent: snom300/8.7.3.10 Allow-Events: dialog X-Real-IP: 192.168.1.106 Supported: path, gruu Authorization: Digest username="2219",realm="asterisk",nonce="60b089fe",uri="sip:192.168.1.84",response="451b5a360de84fc7e662fd2da44fd34d",algorithm=MD5 Content-Length: 0 <-------------> [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: Header 0 [ 33]: REGISTER sip:192.168.1.84 SIP/2.0 [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: Header 1 [ 69]: Via: SIP/2.0/UDP 192.168.1.106:2048;branch=z9hG4bK-byqk1t2m4mx1;rport [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: Header 2 [ 51]: From: "2219" ;tag=klq3amiyhf [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: Header 3 [ 34]: To: "2219" [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: Header 4 [ 34]: Call-ID: 50250e9b9684-8d3g9c7o4hy2 [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: Header 5 [ 17]: CSeq: 30 REGISTER [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: Header 7 [306]: Contact: ;reg-id=1;q=1.0;+sip.instance="";audio;mobility="fixed";duplex="full";description="snom300";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO" [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: Header 8 [ 28]: User-Agent: snom300/8.7.3.10 [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: Header 9 [ 20]: Allow-Events: dialog [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: Header 10 [ 24]: X-Real-IP: 192.168.1.106 [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: Header 11 [ 21]: Supported: path, gruu [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: Header 12 [152]: Authorization: Digest username="2219",realm="asterisk",nonce="60b089fe",uri="sip:192.168.1.84",response="451b5a360de84fc7e662fd2da44fd34d",algorithm=MD5 [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: Header 13 [ 17]: Content-Length: 0 [Aug 10 15:48:45] VERBOSE[17897] chan_sip.c: --- (14 headers 0 lines) --- [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: = Looking for Call ID: 50250e9b9684-8d3g9c7o4hy2 (Checking From) --From tag klq3amiyhf --To-tag [Aug 10 15:48:45] DEBUG[17897] netsock2.c: Splitting '192.168.1.84' into... [Aug 10 15:48:45] DEBUG[17897] netsock2.c: ...host '192.168.1.84' and port ''. [Aug 10 15:48:45] DEBUG[17897] netsock2.c: Splitting '192.168.1.84' into... [Aug 10 15:48:45] DEBUG[17897] netsock2.c: ...host '192.168.1.84' and port ''. [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: Initializing initreq for method REGISTER - callid 50250e9b9684-8d3g9c7o4hy2 [Aug 10 15:48:45] DEBUG[17897] netsock2.c: Splitting '192.168.1.106:2048' into... [Aug 10 15:48:45] DEBUG[17897] netsock2.c: ...host '192.168.1.106' and port '2048'. [Aug 10 15:48:45] VERBOSE[17897] chan_sip.c: Sending to 192.168.1.106:2048 (NAT) [Aug 10 15:48:45] DEBUG[17897] netsock2.c: Splitting '192.168.1.84' into... [Aug 10 15:48:45] DEBUG[17897] netsock2.c: ...host '192.168.1.84' and port ''. [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: Store REGISTER's src-IP:port for call routing. [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: Allocating new SIP dialog for 21a6d07f109fd9cb34513b5b2572d600@127.0.0.1:5060 - OPTIONS (No RTP) [Aug 10 15:48:45] DEBUG[17897] acl.c: For destination '192.168.1.106', our source address is '192.168.1.84'. [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.1.84:5060 [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: Initializing initreq for method OPTIONS - callid 5988d10c1a7958e60da45d762b84836f@192.168.1.84:5060 [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: Header 0 [ 57]: OPTIONS sip:2219@192.168.1.106:2048;line=7g8ggsvj SIP/2.0 [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK436021d5;rport [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: Header 3 [ 59]: From: "asterisk" ;tag=as4c60dc60 [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: Header 4 [ 47]: To: [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: Header 5 [ 41]: Contact: [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: Header 6 [ 59]: Call-ID: 5988d10c1a7958e60da45d762b84836f@192.168.1.84:5060 [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: Header 8 [ 33]: User-Agent: Asterisk PBX 1.8.15.0 [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: Header 9 [ 35]: Date: Fri, 10 Aug 2012 13:48:45 GMT [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 10 15:48:45] VERBOSE[17897] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.106:2048: OPTIONS sip:2219@192.168.1.106:2048;line=7g8ggsvj SIP/2.0 Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK436021d5;rport Max-Forwards: 70 From: "asterisk" ;tag=as4c60dc60 To: Contact: Call-ID: 5988d10c1a7958e60da45d762b84836f@192.168.1.84:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.15.0 Date: Fri, 10 Aug 2012 13:48:45 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #336 [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.1.106:2048 [Aug 10 15:48:45] VERBOSE[17897] chan_sip.c: <--- Transmitting (NAT) to 192.168.1.106:2048 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.106:2048;branch=z9hG4bK-byqk1t2m4mx1;received=192.168.1.106;rport=2048 From: "2219" ;tag=klq3amiyhf To: "2219" ;tag=as0cf23243 Call-ID: 50250e9b9684-8d3g9c7o4hy2 CSeq: 30 REGISTER Server: Asterisk PBX 1.8.15.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Expires: 120 Contact: ;expires=120 Date: Fri, 10 Aug 2012 13:48:45 GMT Content-Length: 0 <------------> [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.1.106:2048 [Aug 10 15:48:45] VERBOSE[17897] chan_sip.c: Scheduling destruction of SIP dialog '50250e9b9684-8d3g9c7o4hy2' in 32000 ms (Method: REGISTER) [Aug 10 15:48:45] DEBUG[17874] devicestate.c: No provider found, checking channel drivers for SIP - 2219 [Aug 10 15:48:45] DEBUG[17874] chan_sip.c: Checking device state for peer 2219 [Aug 10 15:48:45] DEBUG[17874] devicestate.c: Changing state for SIP/2219 - state 1 (Not in use) [Aug 10 15:48:45] DEBUG[17874] devicestate.c: device 'SIP/2219' state '1' [Aug 10 15:48:45] DEBUG[17907] app_queue.c: Device 'SIP/2219' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Aug 10 15:48:45] VERBOSE[17897] chan_sip.c: <--- SIP read from UDP:192.168.1.106:2048 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK436021d5;rport=5060 From: "asterisk" ;tag=as4c60dc60 To: ;tag=ringw8bx5e Call-ID: 5988d10c1a7958e60da45d762b84836f@192.168.1.84:5060 CSeq: 102 OPTIONS Contact: ;reg-id=1 User-Agent: snom300/8.7.3.10 Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: 100rel, replaces, from-change Content-Length: 0 <-------------> [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: Header 1 [ 68]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK436021d5;rport=5060 [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: Header 2 [ 59]: From: "asterisk" ;tag=as4c60dc60 [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: Header 3 [ 62]: To: ;tag=ringw8bx5e [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: Header 4 [ 59]: Call-ID: 5988d10c1a7958e60da45d762b84836f@192.168.1.84:5060 [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: Header 6 [ 61]: Contact: ;reg-id=1 [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: Header 7 [ 28]: User-Agent: snom300/8.7.3.10 [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: Header 8 [ 19]: Accept-Language: en [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: Header 10 [ 96]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: Header 11 [ 42]: Allow-Events: talk, hold, refer, call-info [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: Header 12 [ 40]: Supported: 100rel, replaces, from-change [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: Header 13 [ 17]: Content-Length: 0 [Aug 10 15:48:45] VERBOSE[17897] chan_sip.c: --- (14 headers 0 lines) --- [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: = Looking for Call ID: 5988d10c1a7958e60da45d762b84836f@192.168.1.84:5060 (Checking To) --From tag as4c60dc60 --To-tag ringw8bx5e [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #336 [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: Stopping retransmission on '5988d10c1a7958e60da45d762b84836f@192.168.1.84:5060' of Request 102: Match Found [Aug 10 15:48:45] DEBUG[17897] chan_sip.c: Destroying SIP dialog 5988d10c1a7958e60da45d762b84836f@192.168.1.84:5060 [Aug 10 15:48:45] VERBOSE[17897] chan_sip.c: Really destroying SIP dialog '5988d10c1a7958e60da45d762b84836f@192.168.1.84:5060' Method: OPTIONS [Aug 10 15:48:49] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:48:49] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:48:54] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:48:54] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:48:59] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:48:59] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:49:04] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:49:04] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:49:09] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:49:09] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:49:14] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:49:14] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:49:17] DEBUG[17897] chan_sip.c: Auto destroying SIP dialog '50250e9b9684-8d3g9c7o4hy2' [Aug 10 15:49:17] DEBUG[17897] chan_sip.c: Destroying SIP dialog 50250e9b9684-8d3g9c7o4hy2 [Aug 10 15:49:17] VERBOSE[17897] chan_sip.c: Really destroying SIP dialog '50250e9b9684-8d3g9c7o4hy2' Method: REGISTER [Aug 10 15:49:19] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:49:19] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:49:24] DEBUG[17897] chan_sip.c: Allocating new SIP dialog for 49133c8a2ffb311771a921ff3d36a8bd@127.0.0.1:5060 - OPTIONS (No RTP) [Aug 10 15:49:24] DEBUG[17897] acl.c: For destination '192.168.2.210', our source address is '192.168.1.84'. [Aug 10 15:49:24] DEBUG[17897] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.1.84:5060 [Aug 10 15:49:24] DEBUG[17897] chan_sip.c: Initializing initreq for method OPTIONS - callid 4635f8f35dde75780b9214fa2f189ba1@192.168.1.84:5060 [Aug 10 15:49:24] DEBUG[17897] chan_sip.c: Header 0 [ 43]: OPTIONS sip:2210@192.168.2.210:2048 SIP/2.0 [Aug 10 15:49:24] DEBUG[17897] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK3e390fdc;rport [Aug 10 15:49:24] DEBUG[17897] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 10 15:49:24] DEBUG[17897] chan_sip.c: Header 3 [ 59]: From: "asterisk" ;tag=as31b2a24a [Aug 10 15:49:24] DEBUG[17897] chan_sip.c: Header 4 [ 33]: To: [Aug 10 15:49:24] DEBUG[17897] chan_sip.c: Header 5 [ 41]: Contact: [Aug 10 15:49:24] DEBUG[17897] chan_sip.c: Header 6 [ 59]: Call-ID: 4635f8f35dde75780b9214fa2f189ba1@192.168.1.84:5060 [Aug 10 15:49:24] DEBUG[17897] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:49:24] DEBUG[17897] chan_sip.c: Header 8 [ 33]: User-Agent: Asterisk PBX 1.8.15.0 [Aug 10 15:49:24] DEBUG[17897] chan_sip.c: Header 9 [ 35]: Date: Fri, 10 Aug 2012 13:49:24 GMT [Aug 10 15:49:24] DEBUG[17897] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 10 15:49:24] DEBUG[17897] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 10 15:49:24] VERBOSE[17897] chan_sip.c: Reliably Transmitting (NAT) to 192.168.2.210:2048: OPTIONS sip:2210@192.168.2.210:2048 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK3e390fdc;rport Max-Forwards: 70 From: "asterisk" ;tag=as31b2a24a To: Contact: Call-ID: 4635f8f35dde75780b9214fa2f189ba1@192.168.1.84:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.15.0 Date: Fri, 10 Aug 2012 13:49:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Aug 10 15:49:24] DEBUG[17897] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #340 [Aug 10 15:49:24] DEBUG[17897] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.2.210:2048 [Aug 10 15:49:24] VERBOSE[17897] chan_sip.c: <--- SIP read from UDP:192.168.2.210:2048 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK3e390fdc;rport=5060 From: "asterisk" ;tag=as31b2a24a To: Call-ID: 4635f8f35dde75780b9214fa2f189ba1@192.168.1.84:5060 CSeq: 102 OPTIONS Contact: ;reg-id=1 User-Agent: snom300/8.4.32 Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: 100rel, replaces, from-change Content-Length: 0 <-------------> [Aug 10 15:49:24] DEBUG[17897] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 10 15:49:24] DEBUG[17897] chan_sip.c: Header 1 [ 68]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK3e390fdc;rport=5060 [Aug 10 15:49:24] DEBUG[17897] chan_sip.c: Header 2 [ 59]: From: "asterisk" ;tag=as31b2a24a [Aug 10 15:49:24] DEBUG[17897] chan_sip.c: Header 3 [ 33]: To: [Aug 10 15:49:24] DEBUG[17897] chan_sip.c: Header 4 [ 59]: Call-ID: 4635f8f35dde75780b9214fa2f189ba1@192.168.1.84:5060 [Aug 10 15:49:24] DEBUG[17897] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:49:24] DEBUG[17897] chan_sip.c: Header 6 [ 47]: Contact: ;reg-id=1 [Aug 10 15:49:24] DEBUG[17897] chan_sip.c: Header 7 [ 26]: User-Agent: snom300/8.4.32 [Aug 10 15:49:24] DEBUG[17897] chan_sip.c: Header 8 [ 19]: Accept-Language: en [Aug 10 15:49:24] DEBUG[17897] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [Aug 10 15:49:24] DEBUG[17897] chan_sip.c: Header 10 [ 96]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE [Aug 10 15:49:24] DEBUG[17897] chan_sip.c: Header 11 [ 42]: Allow-Events: talk, hold, refer, call-info [Aug 10 15:49:24] DEBUG[17897] chan_sip.c: Header 12 [ 40]: Supported: 100rel, replaces, from-change [Aug 10 15:49:24] DEBUG[17897] chan_sip.c: Header 13 [ 17]: Content-Length: 0 [Aug 10 15:49:24] VERBOSE[17897] chan_sip.c: --- (14 headers 0 lines) --- [Aug 10 15:49:24] DEBUG[17897] chan_sip.c: = Looking for Call ID: 4635f8f35dde75780b9214fa2f189ba1@192.168.1.84:5060 (Checking To) --From tag as31b2a24a --To-tag [Aug 10 15:49:24] DEBUG[17897] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #340 [Aug 10 15:49:24] DEBUG[17897] chan_sip.c: Stopping retransmission on '4635f8f35dde75780b9214fa2f189ba1@192.168.1.84:5060' of Request 102: Match Found [Aug 10 15:49:24] DEBUG[17897] chan_sip.c: Destroying SIP dialog 4635f8f35dde75780b9214fa2f189ba1@192.168.1.84:5060 [Aug 10 15:49:24] VERBOSE[17897] chan_sip.c: Really destroying SIP dialog '4635f8f35dde75780b9214fa2f189ba1@192.168.1.84:5060' Method: OPTIONS [Aug 10 15:49:24] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:49:24] DEBUG[17897] chan_sip.c: Allocating new SIP dialog for 6210010f0d67f63a3c45d55a09088045@127.0.0.1:5060 - OPTIONS (No RTP) [Aug 10 15:49:24] DEBUG[17897] acl.c: For destination '192.168.1.102', our source address is '192.168.1.84'. [Aug 10 15:49:24] DEBUG[17897] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.1.84:5060 [Aug 10 15:49:24] DEBUG[17897] chan_sip.c: Initializing initreq for method OPTIONS - callid 60b0960f6e695086428e03ba6421e420@192.168.1.84:5060 [Aug 10 15:49:24] DEBUG[17897] chan_sip.c: Header 0 [ 57]: OPTIONS sip:2212@192.168.1.102:2048;line=z1zv7jsj SIP/2.0 [Aug 10 15:49:24] DEBUG[17897] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK4fbd9aec;rport [Aug 10 15:49:24] DEBUG[17897] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 10 15:49:24] DEBUG[17897] chan_sip.c: Header 3 [ 59]: From: "asterisk" ;tag=as1946917a [Aug 10 15:49:24] DEBUG[17897] chan_sip.c: Header 4 [ 47]: To: [Aug 10 15:49:24] DEBUG[17897] chan_sip.c: Header 5 [ 41]: Contact: [Aug 10 15:49:24] DEBUG[17897] chan_sip.c: Header 6 [ 59]: Call-ID: 60b0960f6e695086428e03ba6421e420@192.168.1.84:5060 [Aug 10 15:49:24] DEBUG[17897] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:49:24] DEBUG[17897] chan_sip.c: Header 8 [ 33]: User-Agent: Asterisk PBX 1.8.15.0 [Aug 10 15:49:24] DEBUG[17897] chan_sip.c: Header 9 [ 35]: Date: Fri, 10 Aug 2012 13:49:24 GMT [Aug 10 15:49:24] DEBUG[17897] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 10 15:49:24] DEBUG[17897] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 10 15:49:24] VERBOSE[17897] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.102:2048: OPTIONS sip:2212@192.168.1.102:2048;line=z1zv7jsj SIP/2.0 Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK4fbd9aec;rport Max-Forwards: 70 From: "asterisk" ;tag=as1946917a To: Contact: Call-ID: 60b0960f6e695086428e03ba6421e420@192.168.1.84:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.15.0 Date: Fri, 10 Aug 2012 13:49:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Aug 10 15:49:24] DEBUG[17897] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #343 [Aug 10 15:49:24] DEBUG[17897] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.1.102:2048 [Aug 10 15:49:24] VERBOSE[17897] chan_sip.c: <--- SIP read from UDP:192.168.1.102:2048 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK4fbd9aec;rport=5060 From: "asterisk" ;tag=as1946917a To: Call-ID: 60b0960f6e695086428e03ba6421e420@192.168.1.84:5060 CSeq: 102 OPTIONS Contact: ;reg-id=1 User-Agent: snom300/8.4.32 Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Content-Length: 0 <-------------> [Aug 10 15:49:24] DEBUG[17897] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 10 15:49:24] DEBUG[17897] chan_sip.c: Header 1 [ 68]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK4fbd9aec;rport=5060 [Aug 10 15:49:24] DEBUG[17897] chan_sip.c: Header 2 [ 59]: From: "asterisk" ;tag=as1946917a [Aug 10 15:49:24] DEBUG[17897] chan_sip.c: Header 3 [ 47]: To: [Aug 10 15:49:24] DEBUG[17897] chan_sip.c: Header 4 [ 59]: Call-ID: 60b0960f6e695086428e03ba6421e420@192.168.1.84:5060 [Aug 10 15:49:24] DEBUG[17897] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:49:24] DEBUG[17897] chan_sip.c: Header 6 [ 61]: Contact: ;reg-id=1 [Aug 10 15:49:24] DEBUG[17897] chan_sip.c: Header 7 [ 26]: User-Agent: snom300/8.4.32 [Aug 10 15:49:24] DEBUG[17897] chan_sip.c: Header 8 [ 19]: Accept-Language: en [Aug 10 15:49:24] DEBUG[17897] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [Aug 10 15:49:24] DEBUG[17897] chan_sip.c: Header 10 [ 96]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE [Aug 10 15:49:24] DEBUG[17897] chan_sip.c: Header 11 [ 42]: Allow-Events: talk, hold, refer, call-info [Aug 10 15:49:24] DEBUG[17897] chan_sip.c: Header 12 [ 47]: Supported: timer, 100rel, replaces, from-change [Aug 10 15:49:24] DEBUG[17897] chan_sip.c: Header 13 [ 17]: Content-Length: 0 [Aug 10 15:49:24] VERBOSE[17897] chan_sip.c: --- (14 headers 0 lines) --- [Aug 10 15:49:24] DEBUG[17897] chan_sip.c: = Looking for Call ID: 60b0960f6e695086428e03ba6421e420@192.168.1.84:5060 (Checking To) --From tag as1946917a --To-tag [Aug 10 15:49:24] DEBUG[17897] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #343 [Aug 10 15:49:24] DEBUG[17897] chan_sip.c: Stopping retransmission on '60b0960f6e695086428e03ba6421e420@192.168.1.84:5060' of Request 102: Match Found [Aug 10 15:49:24] DEBUG[17897] chan_sip.c: Destroying SIP dialog 60b0960f6e695086428e03ba6421e420@192.168.1.84:5060 [Aug 10 15:49:24] VERBOSE[17897] chan_sip.c: Really destroying SIP dialog '60b0960f6e695086428e03ba6421e420@192.168.1.84:5060' Method: OPTIONS [Aug 10 15:49:24] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:49:29] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:49:29] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:49:34] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:49:34] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:49:39] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:49:39] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:49:44] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:49:44] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:49:45] DEBUG[17897] chan_sip.c: Allocating new SIP dialog for 7e91313730fa672e480f406430549c57@127.0.0.1:5060 - OPTIONS (No RTP) [Aug 10 15:49:45] DEBUG[17897] acl.c: For destination '192.168.1.106', our source address is '192.168.1.84'. [Aug 10 15:49:45] DEBUG[17897] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.1.84:5060 [Aug 10 15:49:45] DEBUG[17897] chan_sip.c: Initializing initreq for method OPTIONS - callid 18d58cfa0dd6a25b1244a3057df8e2ab@192.168.1.84:5060 [Aug 10 15:49:45] DEBUG[17897] chan_sip.c: Header 0 [ 57]: OPTIONS sip:2219@192.168.1.106:2048;line=7g8ggsvj SIP/2.0 [Aug 10 15:49:45] DEBUG[17897] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK113e41f7;rport [Aug 10 15:49:45] DEBUG[17897] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 10 15:49:45] DEBUG[17897] chan_sip.c: Header 3 [ 59]: From: "asterisk" ;tag=as73d777b6 [Aug 10 15:49:45] DEBUG[17897] chan_sip.c: Header 4 [ 47]: To: [Aug 10 15:49:45] DEBUG[17897] chan_sip.c: Header 5 [ 41]: Contact: [Aug 10 15:49:45] DEBUG[17897] chan_sip.c: Header 6 [ 59]: Call-ID: 18d58cfa0dd6a25b1244a3057df8e2ab@192.168.1.84:5060 [Aug 10 15:49:45] DEBUG[17897] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:49:45] DEBUG[17897] chan_sip.c: Header 8 [ 33]: User-Agent: Asterisk PBX 1.8.15.0 [Aug 10 15:49:45] DEBUG[17897] chan_sip.c: Header 9 [ 35]: Date: Fri, 10 Aug 2012 13:49:45 GMT [Aug 10 15:49:45] DEBUG[17897] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 10 15:49:45] DEBUG[17897] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 10 15:49:45] VERBOSE[17897] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.106:2048: OPTIONS sip:2219@192.168.1.106:2048;line=7g8ggsvj SIP/2.0 Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK113e41f7;rport Max-Forwards: 70 From: "asterisk" ;tag=as73d777b6 To: Contact: Call-ID: 18d58cfa0dd6a25b1244a3057df8e2ab@192.168.1.84:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.15.0 Date: Fri, 10 Aug 2012 13:49:45 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Aug 10 15:49:45] DEBUG[17897] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #346 [Aug 10 15:49:45] DEBUG[17897] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.1.106:2048 [Aug 10 15:49:45] VERBOSE[17897] chan_sip.c: <--- SIP read from UDP:192.168.1.106:2048 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK113e41f7;rport=5060 From: "asterisk" ;tag=as73d777b6 To: ;tag=zhy4zkjz24 Call-ID: 18d58cfa0dd6a25b1244a3057df8e2ab@192.168.1.84:5060 CSeq: 102 OPTIONS Contact: ;reg-id=1 User-Agent: snom300/8.7.3.10 Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: 100rel, replaces, from-change Content-Length: 0 <-------------> [Aug 10 15:49:45] DEBUG[17897] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 10 15:49:45] DEBUG[17897] chan_sip.c: Header 1 [ 68]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK113e41f7;rport=5060 [Aug 10 15:49:45] DEBUG[17897] chan_sip.c: Header 2 [ 59]: From: "asterisk" ;tag=as73d777b6 [Aug 10 15:49:45] DEBUG[17897] chan_sip.c: Header 3 [ 62]: To: ;tag=zhy4zkjz24 [Aug 10 15:49:45] DEBUG[17897] chan_sip.c: Header 4 [ 59]: Call-ID: 18d58cfa0dd6a25b1244a3057df8e2ab@192.168.1.84:5060 [Aug 10 15:49:45] DEBUG[17897] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:49:45] DEBUG[17897] chan_sip.c: Header 6 [ 61]: Contact: ;reg-id=1 [Aug 10 15:49:45] DEBUG[17897] chan_sip.c: Header 7 [ 28]: User-Agent: snom300/8.7.3.10 [Aug 10 15:49:45] DEBUG[17897] chan_sip.c: Header 8 [ 19]: Accept-Language: en [Aug 10 15:49:45] DEBUG[17897] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [Aug 10 15:49:45] DEBUG[17897] chan_sip.c: Header 10 [ 96]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE [Aug 10 15:49:45] DEBUG[17897] chan_sip.c: Header 11 [ 42]: Allow-Events: talk, hold, refer, call-info [Aug 10 15:49:45] DEBUG[17897] chan_sip.c: Header 12 [ 40]: Supported: 100rel, replaces, from-change [Aug 10 15:49:45] DEBUG[17897] chan_sip.c: Header 13 [ 17]: Content-Length: 0 [Aug 10 15:49:45] VERBOSE[17897] chan_sip.c: --- (14 headers 0 lines) --- [Aug 10 15:49:45] DEBUG[17897] chan_sip.c: = Looking for Call ID: 18d58cfa0dd6a25b1244a3057df8e2ab@192.168.1.84:5060 (Checking To) --From tag as73d777b6 --To-tag zhy4zkjz24 [Aug 10 15:49:45] DEBUG[17897] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #346 [Aug 10 15:49:45] DEBUG[17897] chan_sip.c: Stopping retransmission on '18d58cfa0dd6a25b1244a3057df8e2ab@192.168.1.84:5060' of Request 102: Match Found [Aug 10 15:49:45] DEBUG[17897] chan_sip.c: Destroying SIP dialog 18d58cfa0dd6a25b1244a3057df8e2ab@192.168.1.84:5060 [Aug 10 15:49:45] VERBOSE[17897] chan_sip.c: Really destroying SIP dialog '18d58cfa0dd6a25b1244a3057df8e2ab@192.168.1.84:5060' Method: OPTIONS [Aug 10 15:49:45] VERBOSE[17897] chan_sip.c: <--- SIP read from UDP:192.168.1.106:2048 ---> REGISTER sip:192.168.1.84 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.106:2048;branch=z9hG4bK-7se1iclwy1mx;rport From: "2219" ;tag=8t3vagj1cp To: "2219" Call-ID: 50250e9b9684-8d3g9c7o4hy2 CSeq: 31 REGISTER Max-Forwards: 70 Contact: ;reg-id=1;q=1.0;+sip.instance="";audio;mobility="fixed";duplex="full";description="snom300";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO" User-Agent: snom300/8.7.3.10 Allow-Events: dialog X-Real-IP: 192.168.1.106 Supported: path, gruu Content-Length: 0 <-------------> [Aug 10 15:49:45] DEBUG[17897] chan_sip.c: Header 0 [ 33]: REGISTER sip:192.168.1.84 SIP/2.0 [Aug 10 15:49:45] DEBUG[17897] chan_sip.c: Header 1 [ 69]: Via: SIP/2.0/UDP 192.168.1.106:2048;branch=z9hG4bK-7se1iclwy1mx;rport [Aug 10 15:49:45] DEBUG[17897] chan_sip.c: Header 2 [ 51]: From: "2219" ;tag=8t3vagj1cp [Aug 10 15:49:45] DEBUG[17897] chan_sip.c: Header 3 [ 34]: To: "2219" [Aug 10 15:49:45] DEBUG[17897] chan_sip.c: Header 4 [ 34]: Call-ID: 50250e9b9684-8d3g9c7o4hy2 [Aug 10 15:49:45] DEBUG[17897] chan_sip.c: Header 5 [ 17]: CSeq: 31 REGISTER [Aug 10 15:49:45] DEBUG[17897] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Aug 10 15:49:45] DEBUG[17897] chan_sip.c: Header 7 [306]: Contact: ;reg-id=1;q=1.0;+sip.instance="";audio;mobility="fixed";duplex="full";description="snom300";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO" [Aug 10 15:49:45] DEBUG[17897] chan_sip.c: Header 8 [ 28]: User-Agent: snom300/8.7.3.10 [Aug 10 15:49:45] DEBUG[17897] chan_sip.c: Header 9 [ 20]: Allow-Events: dialog [Aug 10 15:49:45] DEBUG[17897] chan_sip.c: Header 10 [ 24]: X-Real-IP: 192.168.1.106 [Aug 10 15:49:45] DEBUG[17897] chan_sip.c: Header 11 [ 21]: Supported: path, gruu [Aug 10 15:49:45] DEBUG[17897] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Aug 10 15:49:45] VERBOSE[17897] chan_sip.c: --- (13 headers 0 lines) --- [Aug 10 15:49:45] DEBUG[17897] chan_sip.c: = Looking for Call ID: 50250e9b9684-8d3g9c7o4hy2 (Checking From) --From tag 8t3vagj1cp --To-tag [Aug 10 15:49:45] DEBUG[17897] acl.c: For destination '192.168.1.106', our source address is '192.168.1.84'. [Aug 10 15:49:45] DEBUG[17897] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.1.84:5060 [Aug 10 15:49:45] DEBUG[17897] chan_sip.c: Allocating new SIP dialog for 50250e9b9684-8d3g9c7o4hy2 - REGISTER (No RTP) [Aug 10 15:49:45] DEBUG[17897] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Aug 10 15:49:45] DEBUG[17897] chan_sip.c: Initializing initreq for method REGISTER - callid 50250e9b9684-8d3g9c7o4hy2 [Aug 10 15:49:45] DEBUG[17897] netsock2.c: Splitting '192.168.1.106:2048' into... [Aug 10 15:49:45] DEBUG[17897] netsock2.c: ...host '192.168.1.106' and port '2048'. [Aug 10 15:49:45] VERBOSE[17897] chan_sip.c: Sending to 192.168.1.106:2048 (NAT) [Aug 10 15:49:45] DEBUG[17897] netsock2.c: Splitting '192.168.1.84' into... [Aug 10 15:49:45] DEBUG[17897] netsock2.c: ...host '192.168.1.84' and port ''. [Aug 10 15:49:45] VERBOSE[17897] chan_sip.c: <--- Transmitting (NAT) to 192.168.1.106:2048 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.106:2048;branch=z9hG4bK-7se1iclwy1mx;received=192.168.1.106;rport=2048 From: "2219" ;tag=8t3vagj1cp To: "2219" ;tag=as40c0f7e3 Call-ID: 50250e9b9684-8d3g9c7o4hy2 CSeq: 31 REGISTER Server: Asterisk PBX 1.8.15.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6cee052d" Content-Length: 0 <------------> [Aug 10 15:49:45] DEBUG[17897] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 192.168.1.106:2048 [Aug 10 15:49:45] VERBOSE[17897] chan_sip.c: Scheduling destruction of SIP dialog '50250e9b9684-8d3g9c7o4hy2' in 32000 ms (Method: REGISTER) [Aug 10 15:49:45] VERBOSE[17897] chan_sip.c: <--- SIP read from UDP:192.168.1.106:2048 ---> REGISTER sip:192.168.1.84 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.106:2048;branch=z9hG4bK-d4f95m138j0t;rport From: "2219" ;tag=8t3vagj1cp To: "2219" Call-ID: 50250e9b9684-8d3g9c7o4hy2 CSeq: 32 REGISTER Max-Forwards: 70 Contact: ;reg-id=1;q=1.0;+sip.instance="";audio;mobility="fixed";duplex="full";description="snom300";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO" User-Agent: snom300/8.7.3.10 Allow-Events: dialog X-Real-IP: 192.168.1.106 Supported: path, gruu Authorization: Digest username="2219",realm="asterisk",nonce="6cee052d",uri="sip:192.168.1.84",response="1c9acfedb211126f5cd95b8a553017a1",algorithm=MD5 Content-Length: 0 <-------------> [Aug 10 15:49:45] DEBUG[17897] chan_sip.c: Header 0 [ 33]: REGISTER sip:192.168.1.84 SIP/2.0 [Aug 10 15:49:45] DEBUG[17897] chan_sip.c: Header 1 [ 69]: Via: SIP/2.0/UDP 192.168.1.106:2048;branch=z9hG4bK-d4f95m138j0t;rport [Aug 10 15:49:45] DEBUG[17897] chan_sip.c: Header 2 [ 51]: From: "2219" ;tag=8t3vagj1cp [Aug 10 15:49:45] DEBUG[17897] chan_sip.c: Header 3 [ 34]: To: "2219" [Aug 10 15:49:45] DEBUG[17897] chan_sip.c: Header 4 [ 34]: Call-ID: 50250e9b9684-8d3g9c7o4hy2 [Aug 10 15:49:45] DEBUG[17897] chan_sip.c: Header 5 [ 17]: CSeq: 32 REGISTER [Aug 10 15:49:45] DEBUG[17897] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Aug 10 15:49:45] DEBUG[17897] chan_sip.c: Header 7 [306]: Contact: ;reg-id=1;q=1.0;+sip.instance="";audio;mobility="fixed";duplex="full";description="snom300";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO" [Aug 10 15:49:45] DEBUG[17897] chan_sip.c: Header 8 [ 28]: User-Agent: snom300/8.7.3.10 [Aug 10 15:49:45] DEBUG[17897] chan_sip.c: Header 9 [ 20]: Allow-Events: dialog [Aug 10 15:49:45] DEBUG[17897] chan_sip.c: Header 10 [ 24]: X-Real-IP: 192.168.1.106 [Aug 10 15:49:45] DEBUG[17897] chan_sip.c: Header 11 [ 21]: Supported: path, gruu [Aug 10 15:49:45] DEBUG[17897] chan_sip.c: Header 12 [152]: Authorization: Digest username="2219",realm="asterisk",nonce="6cee052d",uri="sip:192.168.1.84",response="1c9acfedb211126f5cd95b8a553017a1",algorithm=MD5 [Aug 10 15:49:45] DEBUG[17897] chan_sip.c: Header 13 [ 17]: Content-Length: 0 [Aug 10 15:49:45] VERBOSE[17897] chan_sip.c: --- (14 headers 0 lines) --- [Aug 10 15:49:45] DEBUG[17897] chan_sip.c: = Looking for Call ID: 50250e9b9684-8d3g9c7o4hy2 (Checking From) --From tag 8t3vagj1cp --To-tag [Aug 10 15:49:45] DEBUG[17897] netsock2.c: Splitting '192.168.1.84' into... [Aug 10 15:49:45] DEBUG[17897] netsock2.c: ...host '192.168.1.84' and port ''. [Aug 10 15:49:45] DEBUG[17897] netsock2.c: Splitting '192.168.1.84' into... [Aug 10 15:49:45] DEBUG[17897] netsock2.c: ...host '192.168.1.84' and port ''. [Aug 10 15:49:45] DEBUG[17897] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Aug 10 15:49:45] DEBUG[17897] chan_sip.c: Initializing initreq for method REGISTER - callid 50250e9b9684-8d3g9c7o4hy2 [Aug 10 15:49:45] DEBUG[17897] netsock2.c: Splitting '192.168.1.106:2048' into... [Aug 10 15:49:45] DEBUG[17897] netsock2.c: ...host '192.168.1.106' and port '2048'. [Aug 10 15:49:45] VERBOSE[17897] chan_sip.c: Sending to 192.168.1.106:2048 (NAT) [Aug 10 15:49:45] DEBUG[17897] netsock2.c: Splitting '192.168.1.84' into... [Aug 10 15:49:45] DEBUG[17897] netsock2.c: ...host '192.168.1.84' and port ''. [Aug 10 15:49:45] DEBUG[17897] chan_sip.c: Store REGISTER's src-IP:port for call routing. [Aug 10 15:49:45] DEBUG[17897] chan_sip.c: Allocating new SIP dialog for 4fdddcc43111d5920f038ec958db35aa@127.0.0.1:5060 - OPTIONS (No RTP) [Aug 10 15:49:45] DEBUG[17897] acl.c: For destination '192.168.1.106', our source address is '192.168.1.84'. [Aug 10 15:49:45] DEBUG[17897] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.1.84:5060 [Aug 10 15:49:45] DEBUG[17897] chan_sip.c: Initializing initreq for method OPTIONS - callid 0ee4fa7e35eb759114822aa15a96830d@192.168.1.84:5060 [Aug 10 15:49:45] DEBUG[17897] chan_sip.c: Header 0 [ 57]: OPTIONS sip:2219@192.168.1.106:2048;line=7g8ggsvj SIP/2.0 [Aug 10 15:49:45] DEBUG[17897] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK495fbc5d;rport [Aug 10 15:49:45] DEBUG[17897] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 10 15:49:45] DEBUG[17897] chan_sip.c: Header 3 [ 59]: From: "asterisk" ;tag=as56aa1d97 [Aug 10 15:49:45] DEBUG[17897] chan_sip.c: Header 4 [ 47]: To: [Aug 10 15:49:45] DEBUG[17897] chan_sip.c: Header 5 [ 41]: Contact: [Aug 10 15:49:45] DEBUG[17897] chan_sip.c: Header 6 [ 59]: Call-ID: 0ee4fa7e35eb759114822aa15a96830d@192.168.1.84:5060 [Aug 10 15:49:45] DEBUG[17897] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:49:45] DEBUG[17897] chan_sip.c: Header 8 [ 33]: User-Agent: Asterisk PBX 1.8.15.0 [Aug 10 15:49:45] DEBUG[17897] chan_sip.c: Header 9 [ 35]: Date: Fri, 10 Aug 2012 13:49:45 GMT [Aug 10 15:49:45] DEBUG[17897] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 10 15:49:45] DEBUG[17897] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 10 15:49:45] VERBOSE[17897] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.106:2048: OPTIONS sip:2219@192.168.1.106:2048;line=7g8ggsvj SIP/2.0 Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK495fbc5d;rport Max-Forwards: 70 From: "asterisk" ;tag=as56aa1d97 To: Contact: Call-ID: 0ee4fa7e35eb759114822aa15a96830d@192.168.1.84:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.15.0 Date: Fri, 10 Aug 2012 13:49:45 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Aug 10 15:49:45] DEBUG[17897] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #351 [Aug 10 15:49:45] DEBUG[17897] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.1.106:2048 [Aug 10 15:49:45] VERBOSE[17897] chan_sip.c: <--- Transmitting (NAT) to 192.168.1.106:2048 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.106:2048;branch=z9hG4bK-d4f95m138j0t;received=192.168.1.106;rport=2048 From: "2219" ;tag=8t3vagj1cp To: "2219" ;tag=as40c0f7e3 Call-ID: 50250e9b9684-8d3g9c7o4hy2 CSeq: 32 REGISTER Server: Asterisk PBX 1.8.15.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Expires: 120 Contact: ;expires=120 Date: Fri, 10 Aug 2012 13:49:45 GMT Content-Length: 0 <------------> [Aug 10 15:49:45] DEBUG[17897] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.1.106:2048 [Aug 10 15:49:45] VERBOSE[17897] chan_sip.c: Scheduling destruction of SIP dialog '50250e9b9684-8d3g9c7o4hy2' in 32000 ms (Method: REGISTER) [Aug 10 15:49:45] DEBUG[17874] devicestate.c: No provider found, checking channel drivers for SIP - 2219 [Aug 10 15:49:45] DEBUG[17874] chan_sip.c: Checking device state for peer 2219 [Aug 10 15:49:45] DEBUG[17874] devicestate.c: Changing state for SIP/2219 - state 1 (Not in use) [Aug 10 15:49:45] DEBUG[17874] devicestate.c: device 'SIP/2219' state '1' [Aug 10 15:49:45] DEBUG[17907] app_queue.c: Device 'SIP/2219' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Aug 10 15:49:46] VERBOSE[17897] chan_sip.c: <--- SIP read from UDP:192.168.1.106:2048 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK495fbc5d;rport=5060 From: "asterisk" ;tag=as56aa1d97 To: ;tag=4iiyjw656y Call-ID: 0ee4fa7e35eb759114822aa15a96830d@192.168.1.84:5060 CSeq: 102 OPTIONS Contact: ;reg-id=1 User-Agent: snom300/8.7.3.10 Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: 100rel, replaces, from-change Content-Length: 0 <-------------> [Aug 10 15:49:46] DEBUG[17897] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 10 15:49:46] DEBUG[17897] chan_sip.c: Header 1 [ 68]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK495fbc5d;rport=5060 [Aug 10 15:49:46] DEBUG[17897] chan_sip.c: Header 2 [ 59]: From: "asterisk" ;tag=as56aa1d97 [Aug 10 15:49:46] DEBUG[17897] chan_sip.c: Header 3 [ 62]: To: ;tag=4iiyjw656y [Aug 10 15:49:46] DEBUG[17897] chan_sip.c: Header 4 [ 59]: Call-ID: 0ee4fa7e35eb759114822aa15a96830d@192.168.1.84:5060 [Aug 10 15:49:46] DEBUG[17897] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:49:46] DEBUG[17897] chan_sip.c: Header 6 [ 61]: Contact: ;reg-id=1 [Aug 10 15:49:46] DEBUG[17897] chan_sip.c: Header 7 [ 28]: User-Agent: snom300/8.7.3.10 [Aug 10 15:49:46] DEBUG[17897] chan_sip.c: Header 8 [ 19]: Accept-Language: en [Aug 10 15:49:46] DEBUG[17897] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [Aug 10 15:49:46] DEBUG[17897] chan_sip.c: Header 10 [ 96]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE [Aug 10 15:49:46] DEBUG[17897] chan_sip.c: Header 11 [ 42]: Allow-Events: talk, hold, refer, call-info [Aug 10 15:49:46] DEBUG[17897] chan_sip.c: Header 12 [ 40]: Supported: 100rel, replaces, from-change [Aug 10 15:49:46] DEBUG[17897] chan_sip.c: Header 13 [ 17]: Content-Length: 0 [Aug 10 15:49:46] VERBOSE[17897] chan_sip.c: --- (14 headers 0 lines) --- [Aug 10 15:49:46] DEBUG[17897] chan_sip.c: = Looking for Call ID: 0ee4fa7e35eb759114822aa15a96830d@192.168.1.84:5060 (Checking To) --From tag as56aa1d97 --To-tag 4iiyjw656y [Aug 10 15:49:46] DEBUG[17897] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #351 [Aug 10 15:49:46] DEBUG[17897] chan_sip.c: Stopping retransmission on '0ee4fa7e35eb759114822aa15a96830d@192.168.1.84:5060' of Request 102: Match Found [Aug 10 15:49:46] DEBUG[17897] chan_sip.c: Destroying SIP dialog 0ee4fa7e35eb759114822aa15a96830d@192.168.1.84:5060 [Aug 10 15:49:46] VERBOSE[17897] chan_sip.c: Really destroying SIP dialog '0ee4fa7e35eb759114822aa15a96830d@192.168.1.84:5060' Method: OPTIONS [Aug 10 15:49:49] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:49:49] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:49:54] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:49:54] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:49:59] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:49:59] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:50:04] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:50:04] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:50:09] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:50:09] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:50:14] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:50:14] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:50:17] DEBUG[17897] chan_sip.c: Auto destroying SIP dialog '50250e9b9684-8d3g9c7o4hy2' [Aug 10 15:50:17] DEBUG[17897] chan_sip.c: Destroying SIP dialog 50250e9b9684-8d3g9c7o4hy2 [Aug 10 15:50:17] VERBOSE[17897] chan_sip.c: Really destroying SIP dialog '50250e9b9684-8d3g9c7o4hy2' Method: REGISTER [Aug 10 15:50:19] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:50:19] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:50:24] DEBUG[17897] chan_sip.c: Allocating new SIP dialog for 5610c5da65725c2d4fd8e4be286fc3de@127.0.0.1:5060 - OPTIONS (No RTP) [Aug 10 15:50:24] DEBUG[17897] acl.c: For destination '192.168.2.210', our source address is '192.168.1.84'. [Aug 10 15:50:24] DEBUG[17897] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.1.84:5060 [Aug 10 15:50:24] DEBUG[17897] chan_sip.c: Initializing initreq for method OPTIONS - callid 660aa621253749a957614f5071454827@192.168.1.84:5060 [Aug 10 15:50:24] DEBUG[17897] chan_sip.c: Header 0 [ 43]: OPTIONS sip:2210@192.168.2.210:2048 SIP/2.0 [Aug 10 15:50:24] DEBUG[17897] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK4e277a34;rport [Aug 10 15:50:24] DEBUG[17897] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 10 15:50:24] DEBUG[17897] chan_sip.c: Header 3 [ 59]: From: "asterisk" ;tag=as6105db1d [Aug 10 15:50:24] DEBUG[17897] chan_sip.c: Header 4 [ 33]: To: [Aug 10 15:50:24] DEBUG[17897] chan_sip.c: Header 5 [ 41]: Contact: [Aug 10 15:50:24] DEBUG[17897] chan_sip.c: Header 6 [ 59]: Call-ID: 660aa621253749a957614f5071454827@192.168.1.84:5060 [Aug 10 15:50:24] DEBUG[17897] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:50:24] DEBUG[17897] chan_sip.c: Header 8 [ 33]: User-Agent: Asterisk PBX 1.8.15.0 [Aug 10 15:50:24] DEBUG[17897] chan_sip.c: Header 9 [ 35]: Date: Fri, 10 Aug 2012 13:50:24 GMT [Aug 10 15:50:24] DEBUG[17897] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 10 15:50:24] DEBUG[17897] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 10 15:50:24] VERBOSE[17897] chan_sip.c: Reliably Transmitting (NAT) to 192.168.2.210:2048: OPTIONS sip:2210@192.168.2.210:2048 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK4e277a34;rport Max-Forwards: 70 From: "asterisk" ;tag=as6105db1d To: Contact: Call-ID: 660aa621253749a957614f5071454827@192.168.1.84:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.15.0 Date: Fri, 10 Aug 2012 13:50:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Aug 10 15:50:24] DEBUG[17897] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #355 [Aug 10 15:50:24] DEBUG[17897] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.2.210:2048 [Aug 10 15:50:24] VERBOSE[17897] chan_sip.c: <--- SIP read from UDP:192.168.2.210:2048 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK4e277a34;rport=5060 From: "asterisk" ;tag=as6105db1d To: Call-ID: 660aa621253749a957614f5071454827@192.168.1.84:5060 CSeq: 102 OPTIONS Contact: ;reg-id=1 User-Agent: snom300/8.4.32 Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: 100rel, replaces, from-change Content-Length: 0 <-------------> [Aug 10 15:50:24] DEBUG[17897] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 10 15:50:24] DEBUG[17897] chan_sip.c: Header 1 [ 68]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK4e277a34;rport=5060 [Aug 10 15:50:24] DEBUG[17897] chan_sip.c: Header 2 [ 59]: From: "asterisk" ;tag=as6105db1d [Aug 10 15:50:24] DEBUG[17897] chan_sip.c: Header 3 [ 33]: To: [Aug 10 15:50:24] DEBUG[17897] chan_sip.c: Header 4 [ 59]: Call-ID: 660aa621253749a957614f5071454827@192.168.1.84:5060 [Aug 10 15:50:24] DEBUG[17897] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:50:24] DEBUG[17897] chan_sip.c: Header 6 [ 47]: Contact: ;reg-id=1 [Aug 10 15:50:24] DEBUG[17897] chan_sip.c: Header 7 [ 26]: User-Agent: snom300/8.4.32 [Aug 10 15:50:24] DEBUG[17897] chan_sip.c: Header 8 [ 19]: Accept-Language: en [Aug 10 15:50:24] DEBUG[17897] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [Aug 10 15:50:24] DEBUG[17897] chan_sip.c: Header 10 [ 96]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE [Aug 10 15:50:24] DEBUG[17897] chan_sip.c: Header 11 [ 42]: Allow-Events: talk, hold, refer, call-info [Aug 10 15:50:24] DEBUG[17897] chan_sip.c: Header 12 [ 40]: Supported: 100rel, replaces, from-change [Aug 10 15:50:24] DEBUG[17897] chan_sip.c: Header 13 [ 17]: Content-Length: 0 [Aug 10 15:50:24] VERBOSE[17897] chan_sip.c: --- (14 headers 0 lines) --- [Aug 10 15:50:24] DEBUG[17897] chan_sip.c: = Looking for Call ID: 660aa621253749a957614f5071454827@192.168.1.84:5060 (Checking To) --From tag as6105db1d --To-tag [Aug 10 15:50:24] DEBUG[17897] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #355 [Aug 10 15:50:24] DEBUG[17897] chan_sip.c: Stopping retransmission on '660aa621253749a957614f5071454827@192.168.1.84:5060' of Request 102: Match Found [Aug 10 15:50:24] DEBUG[17897] chan_sip.c: Destroying SIP dialog 660aa621253749a957614f5071454827@192.168.1.84:5060 [Aug 10 15:50:24] VERBOSE[17897] chan_sip.c: Really destroying SIP dialog '660aa621253749a957614f5071454827@192.168.1.84:5060' Method: OPTIONS [Aug 10 15:50:24] DEBUG[17897] chan_sip.c: Allocating new SIP dialog for 2fafae64773dbde9559ce9581a8e5c3d@127.0.0.1:5060 - OPTIONS (No RTP) [Aug 10 15:50:24] DEBUG[17897] acl.c: For destination '192.168.1.102', our source address is '192.168.1.84'. [Aug 10 15:50:24] DEBUG[17897] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.1.84:5060 [Aug 10 15:50:24] DEBUG[17897] chan_sip.c: Initializing initreq for method OPTIONS - callid 03bbebe61403b092668c98f7297e8384@192.168.1.84:5060 [Aug 10 15:50:24] DEBUG[17897] chan_sip.c: Header 0 [ 57]: OPTIONS sip:2212@192.168.1.102:2048;line=z1zv7jsj SIP/2.0 [Aug 10 15:50:24] DEBUG[17897] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK435218ad;rport [Aug 10 15:50:24] DEBUG[17897] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 10 15:50:24] DEBUG[17897] chan_sip.c: Header 3 [ 59]: From: "asterisk" ;tag=as44284e0c [Aug 10 15:50:24] DEBUG[17897] chan_sip.c: Header 4 [ 47]: To: [Aug 10 15:50:24] DEBUG[17897] chan_sip.c: Header 5 [ 41]: Contact: [Aug 10 15:50:24] DEBUG[17897] chan_sip.c: Header 6 [ 59]: Call-ID: 03bbebe61403b092668c98f7297e8384@192.168.1.84:5060 [Aug 10 15:50:24] DEBUG[17897] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:50:24] DEBUG[17897] chan_sip.c: Header 8 [ 33]: User-Agent: Asterisk PBX 1.8.15.0 [Aug 10 15:50:24] DEBUG[17897] chan_sip.c: Header 9 [ 35]: Date: Fri, 10 Aug 2012 13:50:24 GMT [Aug 10 15:50:24] DEBUG[17897] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 10 15:50:24] DEBUG[17897] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 10 15:50:24] VERBOSE[17897] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.102:2048: OPTIONS sip:2212@192.168.1.102:2048;line=z1zv7jsj SIP/2.0 Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK435218ad;rport Max-Forwards: 70 From: "asterisk" ;tag=as44284e0c To: Contact: Call-ID: 03bbebe61403b092668c98f7297e8384@192.168.1.84:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.15.0 Date: Fri, 10 Aug 2012 13:50:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Aug 10 15:50:24] DEBUG[17897] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #358 [Aug 10 15:50:24] DEBUG[17897] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.1.102:2048 [Aug 10 15:50:24] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:50:24] VERBOSE[17897] chan_sip.c: <--- SIP read from UDP:192.168.1.102:2048 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK435218ad;rport=5060 From: "asterisk" ;tag=as44284e0c To: Call-ID: 03bbebe61403b092668c98f7297e8384@192.168.1.84:5060 CSeq: 102 OPTIONS Contact: ;reg-id=1 User-Agent: snom300/8.4.32 Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Content-Length: 0 <-------------> [Aug 10 15:50:24] DEBUG[17897] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 10 15:50:24] DEBUG[17897] chan_sip.c: Header 1 [ 68]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK435218ad;rport=5060 [Aug 10 15:50:24] DEBUG[17897] chan_sip.c: Header 2 [ 59]: From: "asterisk" ;tag=as44284e0c [Aug 10 15:50:24] DEBUG[17897] chan_sip.c: Header 3 [ 47]: To: [Aug 10 15:50:24] DEBUG[17897] chan_sip.c: Header 4 [ 59]: Call-ID: 03bbebe61403b092668c98f7297e8384@192.168.1.84:5060 [Aug 10 15:50:24] DEBUG[17897] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:50:24] DEBUG[17897] chan_sip.c: Header 6 [ 61]: Contact: ;reg-id=1 [Aug 10 15:50:24] DEBUG[17897] chan_sip.c: Header 7 [ 26]: User-Agent: snom300/8.4.32 [Aug 10 15:50:24] DEBUG[17897] chan_sip.c: Header 8 [ 19]: Accept-Language: en [Aug 10 15:50:24] DEBUG[17897] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [Aug 10 15:50:24] DEBUG[17897] chan_sip.c: Header 10 [ 96]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE [Aug 10 15:50:24] DEBUG[17897] chan_sip.c: Header 11 [ 42]: Allow-Events: talk, hold, refer, call-info [Aug 10 15:50:24] DEBUG[17897] chan_sip.c: Header 12 [ 47]: Supported: timer, 100rel, replaces, from-change [Aug 10 15:50:24] DEBUG[17897] chan_sip.c: Header 13 [ 17]: Content-Length: 0 [Aug 10 15:50:24] VERBOSE[17897] chan_sip.c: --- (14 headers 0 lines) --- [Aug 10 15:50:24] DEBUG[17897] chan_sip.c: = Looking for Call ID: 03bbebe61403b092668c98f7297e8384@192.168.1.84:5060 (Checking To) --From tag as44284e0c --To-tag [Aug 10 15:50:24] DEBUG[17897] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #358 [Aug 10 15:50:24] DEBUG[17897] chan_sip.c: Stopping retransmission on '03bbebe61403b092668c98f7297e8384@192.168.1.84:5060' of Request 102: Match Found [Aug 10 15:50:24] DEBUG[17897] chan_sip.c: Destroying SIP dialog 03bbebe61403b092668c98f7297e8384@192.168.1.84:5060 [Aug 10 15:50:24] VERBOSE[17897] chan_sip.c: Really destroying SIP dialog '03bbebe61403b092668c98f7297e8384@192.168.1.84:5060' Method: OPTIONS [Aug 10 15:50:24] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:50:29] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:50:29] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:50:34] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:50:34] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:50:39] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:50:39] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:50:44] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:50:44] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: Allocating new SIP dialog for 6f9d405f1371945776779f310a7ada1a@127.0.0.1:5060 - OPTIONS (No RTP) [Aug 10 15:50:46] DEBUG[17897] acl.c: For destination '192.168.1.106', our source address is '192.168.1.84'. [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.1.84:5060 [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: Initializing initreq for method OPTIONS - callid 5b65f58018c0112b6ed055ac0f2ec02c@192.168.1.84:5060 [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: Header 0 [ 57]: OPTIONS sip:2219@192.168.1.106:2048;line=7g8ggsvj SIP/2.0 [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK6d72ff08;rport [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: Header 3 [ 59]: From: "asterisk" ;tag=as5540710e [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: Header 4 [ 47]: To: [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: Header 5 [ 41]: Contact: [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: Header 6 [ 59]: Call-ID: 5b65f58018c0112b6ed055ac0f2ec02c@192.168.1.84:5060 [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: Header 8 [ 33]: User-Agent: Asterisk PBX 1.8.15.0 [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: Header 9 [ 35]: Date: Fri, 10 Aug 2012 13:50:46 GMT [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 10 15:50:46] VERBOSE[17897] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.106:2048: OPTIONS sip:2219@192.168.1.106:2048;line=7g8ggsvj SIP/2.0 Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK6d72ff08;rport Max-Forwards: 70 From: "asterisk" ;tag=as5540710e To: Contact: Call-ID: 5b65f58018c0112b6ed055ac0f2ec02c@192.168.1.84:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.15.0 Date: Fri, 10 Aug 2012 13:50:46 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #361 [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.1.106:2048 [Aug 10 15:50:46] VERBOSE[17897] chan_sip.c: <--- SIP read from UDP:192.168.1.106:2048 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK6d72ff08;rport=5060 From: "asterisk" ;tag=as5540710e To: ;tag=k8jly8lesw Call-ID: 5b65f58018c0112b6ed055ac0f2ec02c@192.168.1.84:5060 CSeq: 102 OPTIONS Contact: ;reg-id=1 User-Agent: snom300/8.7.3.10 Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: 100rel, replaces, from-change Content-Length: 0 <-------------> [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: Header 1 [ 68]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK6d72ff08;rport=5060 [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: Header 2 [ 59]: From: "asterisk" ;tag=as5540710e [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: Header 3 [ 62]: To: ;tag=k8jly8lesw [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: Header 4 [ 59]: Call-ID: 5b65f58018c0112b6ed055ac0f2ec02c@192.168.1.84:5060 [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: Header 6 [ 61]: Contact: ;reg-id=1 [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: Header 7 [ 28]: User-Agent: snom300/8.7.3.10 [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: Header 8 [ 19]: Accept-Language: en [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: Header 10 [ 96]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: Header 11 [ 42]: Allow-Events: talk, hold, refer, call-info [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: Header 12 [ 40]: Supported: 100rel, replaces, from-change [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: Header 13 [ 17]: Content-Length: 0 [Aug 10 15:50:46] VERBOSE[17897] chan_sip.c: --- (14 headers 0 lines) --- [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: = Looking for Call ID: 5b65f58018c0112b6ed055ac0f2ec02c@192.168.1.84:5060 (Checking To) --From tag as5540710e --To-tag k8jly8lesw [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #361 [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: Stopping retransmission on '5b65f58018c0112b6ed055ac0f2ec02c@192.168.1.84:5060' of Request 102: Match Found [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: Destroying SIP dialog 5b65f58018c0112b6ed055ac0f2ec02c@192.168.1.84:5060 [Aug 10 15:50:46] VERBOSE[17897] chan_sip.c: Really destroying SIP dialog '5b65f58018c0112b6ed055ac0f2ec02c@192.168.1.84:5060' Method: OPTIONS [Aug 10 15:50:46] VERBOSE[17897] chan_sip.c: <--- SIP read from UDP:192.168.1.106:2048 ---> REGISTER sip:192.168.1.84 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.106:2048;branch=z9hG4bK-ied02tm3pe0q;rport From: "2219" ;tag=0cviauvx02 To: "2219" Call-ID: 50250e9b9684-8d3g9c7o4hy2 CSeq: 33 REGISTER Max-Forwards: 70 Contact: ;reg-id=1;q=1.0;+sip.instance="";audio;mobility="fixed";duplex="full";description="snom300";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO" User-Agent: snom300/8.7.3.10 Allow-Events: dialog X-Real-IP: 192.168.1.106 Supported: path, gruu Content-Length: 0 <-------------> [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: Header 0 [ 33]: REGISTER sip:192.168.1.84 SIP/2.0 [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: Header 1 [ 69]: Via: SIP/2.0/UDP 192.168.1.106:2048;branch=z9hG4bK-ied02tm3pe0q;rport [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: Header 2 [ 51]: From: "2219" ;tag=0cviauvx02 [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: Header 3 [ 34]: To: "2219" [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: Header 4 [ 34]: Call-ID: 50250e9b9684-8d3g9c7o4hy2 [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: Header 5 [ 17]: CSeq: 33 REGISTER [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: Header 7 [306]: Contact: ;reg-id=1;q=1.0;+sip.instance="";audio;mobility="fixed";duplex="full";description="snom300";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO" [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: Header 8 [ 28]: User-Agent: snom300/8.7.3.10 [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: Header 9 [ 20]: Allow-Events: dialog [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: Header 10 [ 24]: X-Real-IP: 192.168.1.106 [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: Header 11 [ 21]: Supported: path, gruu [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Aug 10 15:50:46] VERBOSE[17897] chan_sip.c: --- (13 headers 0 lines) --- [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: = Looking for Call ID: 50250e9b9684-8d3g9c7o4hy2 (Checking From) --From tag 0cviauvx02 --To-tag [Aug 10 15:50:46] DEBUG[17897] acl.c: For destination '192.168.1.106', our source address is '192.168.1.84'. [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.1.84:5060 [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: Allocating new SIP dialog for 50250e9b9684-8d3g9c7o4hy2 - REGISTER (No RTP) [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: Initializing initreq for method REGISTER - callid 50250e9b9684-8d3g9c7o4hy2 [Aug 10 15:50:46] DEBUG[17897] netsock2.c: Splitting '192.168.1.106:2048' into... [Aug 10 15:50:46] DEBUG[17897] netsock2.c: ...host '192.168.1.106' and port '2048'. [Aug 10 15:50:46] VERBOSE[17897] chan_sip.c: Sending to 192.168.1.106:2048 (NAT) [Aug 10 15:50:46] DEBUG[17897] netsock2.c: Splitting '192.168.1.84' into... [Aug 10 15:50:46] DEBUG[17897] netsock2.c: ...host '192.168.1.84' and port ''. [Aug 10 15:50:46] VERBOSE[17897] chan_sip.c: <--- Transmitting (NAT) to 192.168.1.106:2048 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.106:2048;branch=z9hG4bK-ied02tm3pe0q;received=192.168.1.106;rport=2048 From: "2219" ;tag=0cviauvx02 To: "2219" ;tag=as5b0cf0d7 Call-ID: 50250e9b9684-8d3g9c7o4hy2 CSeq: 33 REGISTER Server: Asterisk PBX 1.8.15.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7d674cda" Content-Length: 0 <------------> [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 192.168.1.106:2048 [Aug 10 15:50:46] VERBOSE[17897] chan_sip.c: Scheduling destruction of SIP dialog '50250e9b9684-8d3g9c7o4hy2' in 32000 ms (Method: REGISTER) [Aug 10 15:50:46] VERBOSE[17897] chan_sip.c: <--- SIP read from UDP:192.168.1.106:2048 ---> REGISTER sip:192.168.1.84 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.106:2048;branch=z9hG4bK-j1i7r0lok8h6;rport From: "2219" ;tag=0cviauvx02 To: "2219" Call-ID: 50250e9b9684-8d3g9c7o4hy2 CSeq: 34 REGISTER Max-Forwards: 70 Contact: ;reg-id=1;q=1.0;+sip.instance="";audio;mobility="fixed";duplex="full";description="snom300";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO" User-Agent: snom300/8.7.3.10 Allow-Events: dialog X-Real-IP: 192.168.1.106 Supported: path, gruu Authorization: Digest username="2219",realm="asterisk",nonce="7d674cda",uri="sip:192.168.1.84",response="03c7553423fc01da454e9db0745fc8d6",algorithm=MD5 Content-Length: 0 <-------------> [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: Header 0 [ 33]: REGISTER sip:192.168.1.84 SIP/2.0 [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: Header 1 [ 69]: Via: SIP/2.0/UDP 192.168.1.106:2048;branch=z9hG4bK-j1i7r0lok8h6;rport [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: Header 2 [ 51]: From: "2219" ;tag=0cviauvx02 [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: Header 3 [ 34]: To: "2219" [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: Header 4 [ 34]: Call-ID: 50250e9b9684-8d3g9c7o4hy2 [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: Header 5 [ 17]: CSeq: 34 REGISTER [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: Header 7 [306]: Contact: ;reg-id=1;q=1.0;+sip.instance="";audio;mobility="fixed";duplex="full";description="snom300";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO" [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: Header 8 [ 28]: User-Agent: snom300/8.7.3.10 [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: Header 9 [ 20]: Allow-Events: dialog [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: Header 10 [ 24]: X-Real-IP: 192.168.1.106 [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: Header 11 [ 21]: Supported: path, gruu [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: Header 12 [152]: Authorization: Digest username="2219",realm="asterisk",nonce="7d674cda",uri="sip:192.168.1.84",response="03c7553423fc01da454e9db0745fc8d6",algorithm=MD5 [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: Header 13 [ 17]: Content-Length: 0 [Aug 10 15:50:46] VERBOSE[17897] chan_sip.c: --- (14 headers 0 lines) --- [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: = Looking for Call ID: 50250e9b9684-8d3g9c7o4hy2 (Checking From) --From tag 0cviauvx02 --To-tag [Aug 10 15:50:46] DEBUG[17897] netsock2.c: Splitting '192.168.1.84' into... [Aug 10 15:50:46] DEBUG[17897] netsock2.c: ...host '192.168.1.84' and port ''. [Aug 10 15:50:46] DEBUG[17897] netsock2.c: Splitting '192.168.1.84' into... [Aug 10 15:50:46] DEBUG[17897] netsock2.c: ...host '192.168.1.84' and port ''. [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: Initializing initreq for method REGISTER - callid 50250e9b9684-8d3g9c7o4hy2 [Aug 10 15:50:46] DEBUG[17897] netsock2.c: Splitting '192.168.1.106:2048' into... [Aug 10 15:50:46] DEBUG[17897] netsock2.c: ...host '192.168.1.106' and port '2048'. [Aug 10 15:50:46] VERBOSE[17897] chan_sip.c: Sending to 192.168.1.106:2048 (NAT) [Aug 10 15:50:46] DEBUG[17897] netsock2.c: Splitting '192.168.1.84' into... [Aug 10 15:50:46] DEBUG[17897] netsock2.c: ...host '192.168.1.84' and port ''. [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: Store REGISTER's src-IP:port for call routing. [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: Allocating new SIP dialog for 62cca0916758aab755517aa01bb07369@127.0.0.1:5060 - OPTIONS (No RTP) [Aug 10 15:50:46] DEBUG[17897] acl.c: For destination '192.168.1.106', our source address is '192.168.1.84'. [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.1.84:5060 [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: Initializing initreq for method OPTIONS - callid 09f010f012a103771aad551e3853c125@192.168.1.84:5060 [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: Header 0 [ 57]: OPTIONS sip:2219@192.168.1.106:2048;line=7g8ggsvj SIP/2.0 [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK6f271e2f;rport [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: Header 3 [ 59]: From: "asterisk" ;tag=as05e1c344 [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: Header 4 [ 47]: To: [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: Header 5 [ 41]: Contact: [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: Header 6 [ 59]: Call-ID: 09f010f012a103771aad551e3853c125@192.168.1.84:5060 [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: Header 8 [ 33]: User-Agent: Asterisk PBX 1.8.15.0 [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: Header 9 [ 35]: Date: Fri, 10 Aug 2012 13:50:46 GMT [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 10 15:50:46] VERBOSE[17897] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.106:2048: OPTIONS sip:2219@192.168.1.106:2048;line=7g8ggsvj SIP/2.0 Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK6f271e2f;rport Max-Forwards: 70 From: "asterisk" ;tag=as05e1c344 To: Contact: Call-ID: 09f010f012a103771aad551e3853c125@192.168.1.84:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.15.0 Date: Fri, 10 Aug 2012 13:50:46 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #366 [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.1.106:2048 [Aug 10 15:50:46] VERBOSE[17897] chan_sip.c: <--- Transmitting (NAT) to 192.168.1.106:2048 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.106:2048;branch=z9hG4bK-j1i7r0lok8h6;received=192.168.1.106;rport=2048 From: "2219" ;tag=0cviauvx02 To: "2219" ;tag=as5b0cf0d7 Call-ID: 50250e9b9684-8d3g9c7o4hy2 CSeq: 34 REGISTER Server: Asterisk PBX 1.8.15.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Expires: 120 Contact: ;expires=120 Date: Fri, 10 Aug 2012 13:50:46 GMT Content-Length: 0 <------------> [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.1.106:2048 [Aug 10 15:50:46] VERBOSE[17897] chan_sip.c: Scheduling destruction of SIP dialog '50250e9b9684-8d3g9c7o4hy2' in 32000 ms (Method: REGISTER) [Aug 10 15:50:46] DEBUG[17874] devicestate.c: No provider found, checking channel drivers for SIP - 2219 [Aug 10 15:50:46] DEBUG[17874] chan_sip.c: Checking device state for peer 2219 [Aug 10 15:50:46] DEBUG[17874] devicestate.c: Changing state for SIP/2219 - state 1 (Not in use) [Aug 10 15:50:46] DEBUG[17874] devicestate.c: device 'SIP/2219' state '1' [Aug 10 15:50:46] DEBUG[17907] app_queue.c: Device 'SIP/2219' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Aug 10 15:50:46] VERBOSE[17897] chan_sip.c: <--- SIP read from UDP:192.168.1.106:2048 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK6f271e2f;rport=5060 From: "asterisk" ;tag=as05e1c344 To: ;tag=f39r5b3scn Call-ID: 09f010f012a103771aad551e3853c125@192.168.1.84:5060 CSeq: 102 OPTIONS Contact: ;reg-id=1 User-Agent: snom300/8.7.3.10 Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: 100rel, replaces, from-change Content-Length: 0 <-------------> [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: Header 1 [ 68]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK6f271e2f;rport=5060 [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: Header 2 [ 59]: From: "asterisk" ;tag=as05e1c344 [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: Header 3 [ 62]: To: ;tag=f39r5b3scn [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: Header 4 [ 59]: Call-ID: 09f010f012a103771aad551e3853c125@192.168.1.84:5060 [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: Header 6 [ 61]: Contact: ;reg-id=1 [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: Header 7 [ 28]: User-Agent: snom300/8.7.3.10 [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: Header 8 [ 19]: Accept-Language: en [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: Header 10 [ 96]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: Header 11 [ 42]: Allow-Events: talk, hold, refer, call-info [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: Header 12 [ 40]: Supported: 100rel, replaces, from-change [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: Header 13 [ 17]: Content-Length: 0 [Aug 10 15:50:46] VERBOSE[17897] chan_sip.c: --- (14 headers 0 lines) --- [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: = Looking for Call ID: 09f010f012a103771aad551e3853c125@192.168.1.84:5060 (Checking To) --From tag as05e1c344 --To-tag f39r5b3scn [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #366 [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: Stopping retransmission on '09f010f012a103771aad551e3853c125@192.168.1.84:5060' of Request 102: Match Found [Aug 10 15:50:46] DEBUG[17897] chan_sip.c: Destroying SIP dialog 09f010f012a103771aad551e3853c125@192.168.1.84:5060 [Aug 10 15:50:46] VERBOSE[17897] chan_sip.c: Really destroying SIP dialog '09f010f012a103771aad551e3853c125@192.168.1.84:5060' Method: OPTIONS [Aug 10 15:50:49] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:50:49] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:50:54] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:50:54] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:50:59] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:50:59] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:51:04] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:51:04] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:51:09] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:51:09] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:51:14] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:51:14] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:51:18] DEBUG[17897] chan_sip.c: Auto destroying SIP dialog '50250e9b9684-8d3g9c7o4hy2' [Aug 10 15:51:18] DEBUG[17897] chan_sip.c: Destroying SIP dialog 50250e9b9684-8d3g9c7o4hy2 [Aug 10 15:51:18] VERBOSE[17897] chan_sip.c: Really destroying SIP dialog '50250e9b9684-8d3g9c7o4hy2' Method: REGISTER [Aug 10 15:51:19] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:51:19] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:51:24] DEBUG[17897] chan_sip.c: Allocating new SIP dialog for 1c399cf95f61130b7463c8c73be49072@127.0.0.1:5060 - OPTIONS (No RTP) [Aug 10 15:51:24] DEBUG[17897] acl.c: For destination '192.168.2.210', our source address is '192.168.1.84'. [Aug 10 15:51:24] DEBUG[17897] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.1.84:5060 [Aug 10 15:51:24] DEBUG[17897] chan_sip.c: Initializing initreq for method OPTIONS - callid 09b7e9f51c6175bf3455c9881929609b@192.168.1.84:5060 [Aug 10 15:51:24] DEBUG[17897] chan_sip.c: Header 0 [ 43]: OPTIONS sip:2210@192.168.2.210:2048 SIP/2.0 [Aug 10 15:51:24] DEBUG[17897] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK03d8d04b;rport [Aug 10 15:51:24] DEBUG[17897] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 10 15:51:24] DEBUG[17897] chan_sip.c: Header 3 [ 59]: From: "asterisk" ;tag=as6a4dc38f [Aug 10 15:51:24] DEBUG[17897] chan_sip.c: Header 4 [ 33]: To: [Aug 10 15:51:24] DEBUG[17897] chan_sip.c: Header 5 [ 41]: Contact: [Aug 10 15:51:24] DEBUG[17897] chan_sip.c: Header 6 [ 59]: Call-ID: 09b7e9f51c6175bf3455c9881929609b@192.168.1.84:5060 [Aug 10 15:51:24] DEBUG[17897] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:51:24] DEBUG[17897] chan_sip.c: Header 8 [ 33]: User-Agent: Asterisk PBX 1.8.15.0 [Aug 10 15:51:24] DEBUG[17897] chan_sip.c: Header 9 [ 35]: Date: Fri, 10 Aug 2012 13:51:24 GMT [Aug 10 15:51:24] DEBUG[17897] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 10 15:51:24] DEBUG[17897] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 10 15:51:24] VERBOSE[17897] chan_sip.c: Reliably Transmitting (NAT) to 192.168.2.210:2048: OPTIONS sip:2210@192.168.2.210:2048 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK03d8d04b;rport Max-Forwards: 70 From: "asterisk" ;tag=as6a4dc38f To: Contact: Call-ID: 09b7e9f51c6175bf3455c9881929609b@192.168.1.84:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.15.0 Date: Fri, 10 Aug 2012 13:51:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Aug 10 15:51:24] DEBUG[17897] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #370 [Aug 10 15:51:24] DEBUG[17897] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.2.210:2048 [Aug 10 15:51:24] VERBOSE[17897] chan_sip.c: <--- SIP read from UDP:192.168.2.210:2048 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK03d8d04b;rport=5060 From: "asterisk" ;tag=as6a4dc38f To: Call-ID: 09b7e9f51c6175bf3455c9881929609b@192.168.1.84:5060 CSeq: 102 OPTIONS Contact: ;reg-id=1 User-Agent: snom300/8.4.32 Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: 100rel, replaces, from-change Content-Length: 0 <-------------> [Aug 10 15:51:24] DEBUG[17897] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 10 15:51:24] DEBUG[17897] chan_sip.c: Header 1 [ 68]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK03d8d04b;rport=5060 [Aug 10 15:51:24] DEBUG[17897] chan_sip.c: Header 2 [ 59]: From: "asterisk" ;tag=as6a4dc38f [Aug 10 15:51:24] DEBUG[17897] chan_sip.c: Header 3 [ 33]: To: [Aug 10 15:51:24] DEBUG[17897] chan_sip.c: Header 4 [ 59]: Call-ID: 09b7e9f51c6175bf3455c9881929609b@192.168.1.84:5060 [Aug 10 15:51:24] DEBUG[17897] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:51:24] DEBUG[17897] chan_sip.c: Header 6 [ 47]: Contact: ;reg-id=1 [Aug 10 15:51:24] DEBUG[17897] chan_sip.c: Header 7 [ 26]: User-Agent: snom300/8.4.32 [Aug 10 15:51:24] DEBUG[17897] chan_sip.c: Header 8 [ 19]: Accept-Language: en [Aug 10 15:51:24] DEBUG[17897] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [Aug 10 15:51:24] DEBUG[17897] chan_sip.c: Header 10 [ 96]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE [Aug 10 15:51:24] DEBUG[17897] chan_sip.c: Header 11 [ 42]: Allow-Events: talk, hold, refer, call-info [Aug 10 15:51:24] DEBUG[17897] chan_sip.c: Header 12 [ 40]: Supported: 100rel, replaces, from-change [Aug 10 15:51:24] DEBUG[17897] chan_sip.c: Header 13 [ 17]: Content-Length: 0 [Aug 10 15:51:24] VERBOSE[17897] chan_sip.c: --- (14 headers 0 lines) --- [Aug 10 15:51:24] DEBUG[17897] chan_sip.c: = Looking for Call ID: 09b7e9f51c6175bf3455c9881929609b@192.168.1.84:5060 (Checking To) --From tag as6a4dc38f --To-tag [Aug 10 15:51:24] DEBUG[17897] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #370 [Aug 10 15:51:24] DEBUG[17897] chan_sip.c: Stopping retransmission on '09b7e9f51c6175bf3455c9881929609b@192.168.1.84:5060' of Request 102: Match Found [Aug 10 15:51:24] DEBUG[17897] chan_sip.c: Destroying SIP dialog 09b7e9f51c6175bf3455c9881929609b@192.168.1.84:5060 [Aug 10 15:51:24] VERBOSE[17897] chan_sip.c: Really destroying SIP dialog '09b7e9f51c6175bf3455c9881929609b@192.168.1.84:5060' Method: OPTIONS [Aug 10 15:51:24] DEBUG[17897] chan_sip.c: Allocating new SIP dialog for 1d1c95e234fec6aa7a4d6f3d2e94964e@127.0.0.1:5060 - OPTIONS (No RTP) [Aug 10 15:51:24] DEBUG[17897] acl.c: For destination '192.168.1.102', our source address is '192.168.1.84'. [Aug 10 15:51:24] DEBUG[17897] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.1.84:5060 [Aug 10 15:51:24] DEBUG[17897] chan_sip.c: Initializing initreq for method OPTIONS - callid 32981ef46fb84fe306ce390e460563bf@192.168.1.84:5060 [Aug 10 15:51:24] DEBUG[17897] chan_sip.c: Header 0 [ 57]: OPTIONS sip:2212@192.168.1.102:2048;line=z1zv7jsj SIP/2.0 [Aug 10 15:51:24] DEBUG[17897] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK75598494;rport [Aug 10 15:51:24] DEBUG[17897] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 10 15:51:24] DEBUG[17897] chan_sip.c: Header 3 [ 59]: From: "asterisk" ;tag=as38eb4048 [Aug 10 15:51:24] DEBUG[17897] chan_sip.c: Header 4 [ 47]: To: [Aug 10 15:51:24] DEBUG[17897] chan_sip.c: Header 5 [ 41]: Contact: [Aug 10 15:51:24] DEBUG[17897] chan_sip.c: Header 6 [ 59]: Call-ID: 32981ef46fb84fe306ce390e460563bf@192.168.1.84:5060 [Aug 10 15:51:24] DEBUG[17897] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:51:24] DEBUG[17897] chan_sip.c: Header 8 [ 33]: User-Agent: Asterisk PBX 1.8.15.0 [Aug 10 15:51:24] DEBUG[17897] chan_sip.c: Header 9 [ 35]: Date: Fri, 10 Aug 2012 13:51:24 GMT [Aug 10 15:51:24] DEBUG[17897] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 10 15:51:24] DEBUG[17897] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 10 15:51:24] VERBOSE[17897] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.102:2048: OPTIONS sip:2212@192.168.1.102:2048;line=z1zv7jsj SIP/2.0 Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK75598494;rport Max-Forwards: 70 From: "asterisk" ;tag=as38eb4048 To: Contact: Call-ID: 32981ef46fb84fe306ce390e460563bf@192.168.1.84:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.15.0 Date: Fri, 10 Aug 2012 13:51:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Aug 10 15:51:24] DEBUG[17897] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #373 [Aug 10 15:51:24] DEBUG[17897] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.1.102:2048 [Aug 10 15:51:24] VERBOSE[17897] chan_sip.c: <--- SIP read from UDP:192.168.1.102:2048 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK75598494;rport=5060 From: "asterisk" ;tag=as38eb4048 To: Call-ID: 32981ef46fb84fe306ce390e460563bf@192.168.1.84:5060 CSeq: 102 OPTIONS Contact: ;reg-id=1 User-Agent: snom300/8.4.32 Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Content-Length: 0 <-------------> [Aug 10 15:51:24] DEBUG[17897] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 10 15:51:24] DEBUG[17897] chan_sip.c: Header 1 [ 68]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK75598494;rport=5060 [Aug 10 15:51:24] DEBUG[17897] chan_sip.c: Header 2 [ 59]: From: "asterisk" ;tag=as38eb4048 [Aug 10 15:51:24] DEBUG[17897] chan_sip.c: Header 3 [ 47]: To: [Aug 10 15:51:24] DEBUG[17897] chan_sip.c: Header 4 [ 59]: Call-ID: 32981ef46fb84fe306ce390e460563bf@192.168.1.84:5060 [Aug 10 15:51:24] DEBUG[17897] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:51:24] DEBUG[17897] chan_sip.c: Header 6 [ 61]: Contact: ;reg-id=1 [Aug 10 15:51:24] DEBUG[17897] chan_sip.c: Header 7 [ 26]: User-Agent: snom300/8.4.32 [Aug 10 15:51:24] DEBUG[17897] chan_sip.c: Header 8 [ 19]: Accept-Language: en [Aug 10 15:51:24] DEBUG[17897] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [Aug 10 15:51:24] DEBUG[17897] chan_sip.c: Header 10 [ 96]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE [Aug 10 15:51:24] DEBUG[17897] chan_sip.c: Header 11 [ 42]: Allow-Events: talk, hold, refer, call-info [Aug 10 15:51:24] DEBUG[17897] chan_sip.c: Header 12 [ 47]: Supported: timer, 100rel, replaces, from-change [Aug 10 15:51:24] DEBUG[17897] chan_sip.c: Header 13 [ 17]: Content-Length: 0 [Aug 10 15:51:24] VERBOSE[17897] chan_sip.c: --- (14 headers 0 lines) --- [Aug 10 15:51:24] DEBUG[17897] chan_sip.c: = Looking for Call ID: 32981ef46fb84fe306ce390e460563bf@192.168.1.84:5060 (Checking To) --From tag as38eb4048 --To-tag [Aug 10 15:51:24] DEBUG[17897] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #373 [Aug 10 15:51:24] DEBUG[17897] chan_sip.c: Stopping retransmission on '32981ef46fb84fe306ce390e460563bf@192.168.1.84:5060' of Request 102: Match Found [Aug 10 15:51:24] DEBUG[17897] chan_sip.c: Destroying SIP dialog 32981ef46fb84fe306ce390e460563bf@192.168.1.84:5060 [Aug 10 15:51:24] VERBOSE[17897] chan_sip.c: Really destroying SIP dialog '32981ef46fb84fe306ce390e460563bf@192.168.1.84:5060' Method: OPTIONS [Aug 10 15:51:24] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:51:24] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:51:29] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:51:29] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:51:34] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:51:34] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:51:39] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:51:39] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:51:44] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:51:44] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: Allocating new SIP dialog for 6aeedf255ed1f4f27c637a77674e35b4@127.0.0.1:5060 - OPTIONS (No RTP) [Aug 10 15:51:46] DEBUG[17897] acl.c: For destination '192.168.1.106', our source address is '192.168.1.84'. [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.1.84:5060 [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: Initializing initreq for method OPTIONS - callid 73da4730584c1c9c7d18fa4a6328fb74@192.168.1.84:5060 [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: Header 0 [ 57]: OPTIONS sip:2219@192.168.1.106:2048;line=7g8ggsvj SIP/2.0 [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK411c3cbf;rport [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: Header 3 [ 59]: From: "asterisk" ;tag=as7c2afffb [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: Header 4 [ 47]: To: [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: Header 5 [ 41]: Contact: [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: Header 6 [ 59]: Call-ID: 73da4730584c1c9c7d18fa4a6328fb74@192.168.1.84:5060 [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: Header 8 [ 33]: User-Agent: Asterisk PBX 1.8.15.0 [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: Header 9 [ 35]: Date: Fri, 10 Aug 2012 13:51:46 GMT [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 10 15:51:46] VERBOSE[17897] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.106:2048: OPTIONS sip:2219@192.168.1.106:2048;line=7g8ggsvj SIP/2.0 Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK411c3cbf;rport Max-Forwards: 70 From: "asterisk" ;tag=as7c2afffb To: Contact: Call-ID: 73da4730584c1c9c7d18fa4a6328fb74@192.168.1.84:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.15.0 Date: Fri, 10 Aug 2012 13:51:46 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #376 [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.1.106:2048 [Aug 10 15:51:46] VERBOSE[17897] chan_sip.c: <--- SIP read from UDP:192.168.1.106:2048 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK411c3cbf;rport=5060 From: "asterisk" ;tag=as7c2afffb To: ;tag=rjtij67sq8 Call-ID: 73da4730584c1c9c7d18fa4a6328fb74@192.168.1.84:5060 CSeq: 102 OPTIONS Contact: ;reg-id=1 User-Agent: snom300/8.7.3.10 Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: 100rel, replaces, from-change Content-Length: 0 <-------------> [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: Header 1 [ 68]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK411c3cbf;rport=5060 [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: Header 2 [ 59]: From: "asterisk" ;tag=as7c2afffb [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: Header 3 [ 62]: To: ;tag=rjtij67sq8 [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: Header 4 [ 59]: Call-ID: 73da4730584c1c9c7d18fa4a6328fb74@192.168.1.84:5060 [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: Header 6 [ 61]: Contact: ;reg-id=1 [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: Header 7 [ 28]: User-Agent: snom300/8.7.3.10 [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: Header 8 [ 19]: Accept-Language: en [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: Header 10 [ 96]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: Header 11 [ 42]: Allow-Events: talk, hold, refer, call-info [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: Header 12 [ 40]: Supported: 100rel, replaces, from-change [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: Header 13 [ 17]: Content-Length: 0 [Aug 10 15:51:46] VERBOSE[17897] chan_sip.c: --- (14 headers 0 lines) --- [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: = Looking for Call ID: 73da4730584c1c9c7d18fa4a6328fb74@192.168.1.84:5060 (Checking To) --From tag as7c2afffb --To-tag rjtij67sq8 [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #376 [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: Stopping retransmission on '73da4730584c1c9c7d18fa4a6328fb74@192.168.1.84:5060' of Request 102: Match Found [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: Destroying SIP dialog 73da4730584c1c9c7d18fa4a6328fb74@192.168.1.84:5060 [Aug 10 15:51:46] VERBOSE[17897] chan_sip.c: Really destroying SIP dialog '73da4730584c1c9c7d18fa4a6328fb74@192.168.1.84:5060' Method: OPTIONS [Aug 10 15:51:46] VERBOSE[17897] chan_sip.c: <--- SIP read from UDP:192.168.1.106:2048 ---> REGISTER sip:192.168.1.84 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.106:2048;branch=z9hG4bK-nmnewjl38atz;rport From: "2219" ;tag=25rrdx3x5v To: "2219" Call-ID: 50250e9b9684-8d3g9c7o4hy2 CSeq: 35 REGISTER Max-Forwards: 70 Contact: ;reg-id=1;q=1.0;+sip.instance="";audio;mobility="fixed";duplex="full";description="snom300";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO" User-Agent: snom300/8.7.3.10 Allow-Events: dialog X-Real-IP: 192.168.1.106 Supported: path, gruu Content-Length: 0 <-------------> [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: Header 0 [ 33]: REGISTER sip:192.168.1.84 SIP/2.0 [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: Header 1 [ 69]: Via: SIP/2.0/UDP 192.168.1.106:2048;branch=z9hG4bK-nmnewjl38atz;rport [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: Header 2 [ 51]: From: "2219" ;tag=25rrdx3x5v [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: Header 3 [ 34]: To: "2219" [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: Header 4 [ 34]: Call-ID: 50250e9b9684-8d3g9c7o4hy2 [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: Header 5 [ 17]: CSeq: 35 REGISTER [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: Header 7 [306]: Contact: ;reg-id=1;q=1.0;+sip.instance="";audio;mobility="fixed";duplex="full";description="snom300";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO" [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: Header 8 [ 28]: User-Agent: snom300/8.7.3.10 [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: Header 9 [ 20]: Allow-Events: dialog [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: Header 10 [ 24]: X-Real-IP: 192.168.1.106 [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: Header 11 [ 21]: Supported: path, gruu [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Aug 10 15:51:46] VERBOSE[17897] chan_sip.c: --- (13 headers 0 lines) --- [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: = Looking for Call ID: 50250e9b9684-8d3g9c7o4hy2 (Checking From) --From tag 25rrdx3x5v --To-tag [Aug 10 15:51:46] DEBUG[17897] acl.c: For destination '192.168.1.106', our source address is '192.168.1.84'. [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.1.84:5060 [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: Allocating new SIP dialog for 50250e9b9684-8d3g9c7o4hy2 - REGISTER (No RTP) [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: Initializing initreq for method REGISTER - callid 50250e9b9684-8d3g9c7o4hy2 [Aug 10 15:51:46] DEBUG[17897] netsock2.c: Splitting '192.168.1.106:2048' into... [Aug 10 15:51:46] DEBUG[17897] netsock2.c: ...host '192.168.1.106' and port '2048'. [Aug 10 15:51:46] VERBOSE[17897] chan_sip.c: Sending to 192.168.1.106:2048 (NAT) [Aug 10 15:51:46] DEBUG[17897] netsock2.c: Splitting '192.168.1.84' into... [Aug 10 15:51:46] DEBUG[17897] netsock2.c: ...host '192.168.1.84' and port ''. [Aug 10 15:51:46] VERBOSE[17897] chan_sip.c: <--- Transmitting (NAT) to 192.168.1.106:2048 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.106:2048;branch=z9hG4bK-nmnewjl38atz;received=192.168.1.106;rport=2048 From: "2219" ;tag=25rrdx3x5v To: "2219" ;tag=as2db0c2ee Call-ID: 50250e9b9684-8d3g9c7o4hy2 CSeq: 35 REGISTER Server: Asterisk PBX 1.8.15.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="46cff508" Content-Length: 0 <------------> [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 192.168.1.106:2048 [Aug 10 15:51:46] VERBOSE[17897] chan_sip.c: Scheduling destruction of SIP dialog '50250e9b9684-8d3g9c7o4hy2' in 32000 ms (Method: REGISTER) [Aug 10 15:51:46] VERBOSE[17897] chan_sip.c: <--- SIP read from UDP:192.168.1.106:2048 ---> REGISTER sip:192.168.1.84 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.106:2048;branch=z9hG4bK-5h6m9t8grpzc;rport From: "2219" ;tag=25rrdx3x5v To: "2219" Call-ID: 50250e9b9684-8d3g9c7o4hy2 CSeq: 36 REGISTER Max-Forwards: 70 Contact: ;reg-id=1;q=1.0;+sip.instance="";audio;mobility="fixed";duplex="full";description="snom300";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO" User-Agent: snom300/8.7.3.10 Allow-Events: dialog X-Real-IP: 192.168.1.106 Supported: path, gruu Authorization: Digest username="2219",realm="asterisk",nonce="46cff508",uri="sip:192.168.1.84",response="9c158ca9087ac22c7277158fe31e84cd",algorithm=MD5 Content-Length: 0 <-------------> [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: Header 0 [ 33]: REGISTER sip:192.168.1.84 SIP/2.0 [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: Header 1 [ 69]: Via: SIP/2.0/UDP 192.168.1.106:2048;branch=z9hG4bK-5h6m9t8grpzc;rport [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: Header 2 [ 51]: From: "2219" ;tag=25rrdx3x5v [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: Header 3 [ 34]: To: "2219" [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: Header 4 [ 34]: Call-ID: 50250e9b9684-8d3g9c7o4hy2 [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: Header 5 [ 17]: CSeq: 36 REGISTER [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: Header 7 [306]: Contact: ;reg-id=1;q=1.0;+sip.instance="";audio;mobility="fixed";duplex="full";description="snom300";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO" [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: Header 8 [ 28]: User-Agent: snom300/8.7.3.10 [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: Header 9 [ 20]: Allow-Events: dialog [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: Header 10 [ 24]: X-Real-IP: 192.168.1.106 [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: Header 11 [ 21]: Supported: path, gruu [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: Header 12 [152]: Authorization: Digest username="2219",realm="asterisk",nonce="46cff508",uri="sip:192.168.1.84",response="9c158ca9087ac22c7277158fe31e84cd",algorithm=MD5 [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: Header 13 [ 17]: Content-Length: 0 [Aug 10 15:51:46] VERBOSE[17897] chan_sip.c: --- (14 headers 0 lines) --- [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: = Looking for Call ID: 50250e9b9684-8d3g9c7o4hy2 (Checking From) --From tag 25rrdx3x5v --To-tag [Aug 10 15:51:46] DEBUG[17897] netsock2.c: Splitting '192.168.1.84' into... [Aug 10 15:51:46] DEBUG[17897] netsock2.c: ...host '192.168.1.84' and port ''. [Aug 10 15:51:46] DEBUG[17897] netsock2.c: Splitting '192.168.1.84' into... [Aug 10 15:51:46] DEBUG[17897] netsock2.c: ...host '192.168.1.84' and port ''. [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: Initializing initreq for method REGISTER - callid 50250e9b9684-8d3g9c7o4hy2 [Aug 10 15:51:46] DEBUG[17897] netsock2.c: Splitting '192.168.1.106:2048' into... [Aug 10 15:51:46] DEBUG[17897] netsock2.c: ...host '192.168.1.106' and port '2048'. [Aug 10 15:51:46] VERBOSE[17897] chan_sip.c: Sending to 192.168.1.106:2048 (NAT) [Aug 10 15:51:46] DEBUG[17897] netsock2.c: Splitting '192.168.1.84' into... [Aug 10 15:51:46] DEBUG[17897] netsock2.c: ...host '192.168.1.84' and port ''. [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: Store REGISTER's src-IP:port for call routing. [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: Allocating new SIP dialog for 411863490f556c2a3992c9d66258b081@127.0.0.1:5060 - OPTIONS (No RTP) [Aug 10 15:51:46] DEBUG[17897] acl.c: For destination '192.168.1.106', our source address is '192.168.1.84'. [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.1.84:5060 [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: Initializing initreq for method OPTIONS - callid 4d4de06872b25a51760c799a53a00b31@192.168.1.84:5060 [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: Header 0 [ 57]: OPTIONS sip:2219@192.168.1.106:2048;line=7g8ggsvj SIP/2.0 [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK6ae8f962;rport [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: Header 3 [ 59]: From: "asterisk" ;tag=as65ab5404 [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: Header 4 [ 47]: To: [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: Header 5 [ 41]: Contact: [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: Header 6 [ 59]: Call-ID: 4d4de06872b25a51760c799a53a00b31@192.168.1.84:5060 [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: Header 8 [ 33]: User-Agent: Asterisk PBX 1.8.15.0 [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: Header 9 [ 35]: Date: Fri, 10 Aug 2012 13:51:46 GMT [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 10 15:51:46] VERBOSE[17897] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.106:2048: OPTIONS sip:2219@192.168.1.106:2048;line=7g8ggsvj SIP/2.0 Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK6ae8f962;rport Max-Forwards: 70 From: "asterisk" ;tag=as65ab5404 To: Contact: Call-ID: 4d4de06872b25a51760c799a53a00b31@192.168.1.84:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.15.0 Date: Fri, 10 Aug 2012 13:51:46 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #381 [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.1.106:2048 [Aug 10 15:51:46] VERBOSE[17897] chan_sip.c: <--- Transmitting (NAT) to 192.168.1.106:2048 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.106:2048;branch=z9hG4bK-5h6m9t8grpzc;received=192.168.1.106;rport=2048 From: "2219" ;tag=25rrdx3x5v To: "2219" ;tag=as2db0c2ee Call-ID: 50250e9b9684-8d3g9c7o4hy2 CSeq: 36 REGISTER Server: Asterisk PBX 1.8.15.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Expires: 120 Contact: ;expires=120 Date: Fri, 10 Aug 2012 13:51:46 GMT Content-Length: 0 <------------> [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.1.106:2048 [Aug 10 15:51:46] VERBOSE[17897] chan_sip.c: Scheduling destruction of SIP dialog '50250e9b9684-8d3g9c7o4hy2' in 32000 ms (Method: REGISTER) [Aug 10 15:51:46] DEBUG[17874] devicestate.c: No provider found, checking channel drivers for SIP - 2219 [Aug 10 15:51:46] DEBUG[17874] chan_sip.c: Checking device state for peer 2219 [Aug 10 15:51:46] DEBUG[17874] devicestate.c: Changing state for SIP/2219 - state 1 (Not in use) [Aug 10 15:51:46] DEBUG[17874] devicestate.c: device 'SIP/2219' state '1' [Aug 10 15:51:46] DEBUG[17907] app_queue.c: Device 'SIP/2219' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Aug 10 15:51:46] VERBOSE[17897] chan_sip.c: <--- SIP read from UDP:192.168.1.106:2048 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK6ae8f962;rport=5060 From: "asterisk" ;tag=as65ab5404 To: ;tag=0ciu3ngtr5 Call-ID: 4d4de06872b25a51760c799a53a00b31@192.168.1.84:5060 CSeq: 102 OPTIONS Contact: ;reg-id=1 User-Agent: snom300/8.7.3.10 Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: 100rel, replaces, from-change Content-Length: 0 <-------------> [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: Header 1 [ 68]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK6ae8f962;rport=5060 [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: Header 2 [ 59]: From: "asterisk" ;tag=as65ab5404 [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: Header 3 [ 62]: To: ;tag=0ciu3ngtr5 [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: Header 4 [ 59]: Call-ID: 4d4de06872b25a51760c799a53a00b31@192.168.1.84:5060 [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: Header 6 [ 61]: Contact: ;reg-id=1 [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: Header 7 [ 28]: User-Agent: snom300/8.7.3.10 [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: Header 8 [ 19]: Accept-Language: en [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: Header 10 [ 96]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: Header 11 [ 42]: Allow-Events: talk, hold, refer, call-info [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: Header 12 [ 40]: Supported: 100rel, replaces, from-change [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: Header 13 [ 17]: Content-Length: 0 [Aug 10 15:51:46] VERBOSE[17897] chan_sip.c: --- (14 headers 0 lines) --- [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: = Looking for Call ID: 4d4de06872b25a51760c799a53a00b31@192.168.1.84:5060 (Checking To) --From tag as65ab5404 --To-tag 0ciu3ngtr5 [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #381 [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: Stopping retransmission on '4d4de06872b25a51760c799a53a00b31@192.168.1.84:5060' of Request 102: Match Found [Aug 10 15:51:46] DEBUG[17897] chan_sip.c: Destroying SIP dialog 4d4de06872b25a51760c799a53a00b31@192.168.1.84:5060 [Aug 10 15:51:46] VERBOSE[17897] chan_sip.c: Really destroying SIP dialog '4d4de06872b25a51760c799a53a00b31@192.168.1.84:5060' Method: OPTIONS [Aug 10 15:51:49] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:51:49] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:51:54] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:51:54] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:51:59] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:51:59] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:52:04] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:52:04] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:52:09] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:52:09] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:52:14] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:52:14] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:52:18] DEBUG[17897] chan_sip.c: Auto destroying SIP dialog '50250e9b9684-8d3g9c7o4hy2' [Aug 10 15:52:18] DEBUG[17897] chan_sip.c: Destroying SIP dialog 50250e9b9684-8d3g9c7o4hy2 [Aug 10 15:52:18] VERBOSE[17897] chan_sip.c: Really destroying SIP dialog '50250e9b9684-8d3g9c7o4hy2' Method: REGISTER [Aug 10 15:52:19] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:52:19] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:52:24] DEBUG[17897] chan_sip.c: Allocating new SIP dialog for 5c21179a7aed3ae02cb2e2640a8017c2@127.0.0.1:5060 - OPTIONS (No RTP) [Aug 10 15:52:24] DEBUG[17897] acl.c: For destination '192.168.2.210', our source address is '192.168.1.84'. [Aug 10 15:52:24] DEBUG[17897] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.1.84:5060 [Aug 10 15:52:24] DEBUG[17897] chan_sip.c: Initializing initreq for method OPTIONS - callid 4b8eeb0d0347d48b7ee94c7a5334f9bb@192.168.1.84:5060 [Aug 10 15:52:24] DEBUG[17897] chan_sip.c: Header 0 [ 43]: OPTIONS sip:2210@192.168.2.210:2048 SIP/2.0 [Aug 10 15:52:24] DEBUG[17897] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK7d71c179;rport [Aug 10 15:52:24] DEBUG[17897] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 10 15:52:24] DEBUG[17897] chan_sip.c: Header 3 [ 59]: From: "asterisk" ;tag=as46033be3 [Aug 10 15:52:24] DEBUG[17897] chan_sip.c: Header 4 [ 33]: To: [Aug 10 15:52:24] DEBUG[17897] chan_sip.c: Header 5 [ 41]: Contact: [Aug 10 15:52:24] DEBUG[17897] chan_sip.c: Header 6 [ 59]: Call-ID: 4b8eeb0d0347d48b7ee94c7a5334f9bb@192.168.1.84:5060 [Aug 10 15:52:24] DEBUG[17897] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:52:24] DEBUG[17897] chan_sip.c: Header 8 [ 33]: User-Agent: Asterisk PBX 1.8.15.0 [Aug 10 15:52:24] DEBUG[17897] chan_sip.c: Header 9 [ 35]: Date: Fri, 10 Aug 2012 13:52:24 GMT [Aug 10 15:52:24] DEBUG[17897] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 10 15:52:24] DEBUG[17897] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 10 15:52:24] VERBOSE[17897] chan_sip.c: Reliably Transmitting (NAT) to 192.168.2.210:2048: OPTIONS sip:2210@192.168.2.210:2048 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK7d71c179;rport Max-Forwards: 70 From: "asterisk" ;tag=as46033be3 To: Contact: Call-ID: 4b8eeb0d0347d48b7ee94c7a5334f9bb@192.168.1.84:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.15.0 Date: Fri, 10 Aug 2012 13:52:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Aug 10 15:52:24] DEBUG[17897] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #385 [Aug 10 15:52:24] DEBUG[17897] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.2.210:2048 [Aug 10 15:52:24] VERBOSE[17897] chan_sip.c: <--- SIP read from UDP:192.168.2.210:2048 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK7d71c179;rport=5060 From: "asterisk" ;tag=as46033be3 To: Call-ID: 4b8eeb0d0347d48b7ee94c7a5334f9bb@192.168.1.84:5060 CSeq: 102 OPTIONS Contact: ;reg-id=1 User-Agent: snom300/8.4.32 Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: 100rel, replaces, from-change Content-Length: 0 <-------------> [Aug 10 15:52:24] DEBUG[17897] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 10 15:52:24] DEBUG[17897] chan_sip.c: Header 1 [ 68]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK7d71c179;rport=5060 [Aug 10 15:52:24] DEBUG[17897] chan_sip.c: Header 2 [ 59]: From: "asterisk" ;tag=as46033be3 [Aug 10 15:52:24] DEBUG[17897] chan_sip.c: Header 3 [ 33]: To: [Aug 10 15:52:24] DEBUG[17897] chan_sip.c: Header 4 [ 59]: Call-ID: 4b8eeb0d0347d48b7ee94c7a5334f9bb@192.168.1.84:5060 [Aug 10 15:52:24] DEBUG[17897] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:52:24] DEBUG[17897] chan_sip.c: Header 6 [ 47]: Contact: ;reg-id=1 [Aug 10 15:52:24] DEBUG[17897] chan_sip.c: Header 7 [ 26]: User-Agent: snom300/8.4.32 [Aug 10 15:52:24] DEBUG[17897] chan_sip.c: Header 8 [ 19]: Accept-Language: en [Aug 10 15:52:24] DEBUG[17897] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [Aug 10 15:52:24] DEBUG[17897] chan_sip.c: Header 10 [ 96]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE [Aug 10 15:52:24] DEBUG[17897] chan_sip.c: Header 11 [ 42]: Allow-Events: talk, hold, refer, call-info [Aug 10 15:52:24] DEBUG[17897] chan_sip.c: Header 12 [ 40]: Supported: 100rel, replaces, from-change [Aug 10 15:52:24] DEBUG[17897] chan_sip.c: Header 13 [ 17]: Content-Length: 0 [Aug 10 15:52:24] VERBOSE[17897] chan_sip.c: --- (14 headers 0 lines) --- [Aug 10 15:52:24] DEBUG[17897] chan_sip.c: = Looking for Call ID: 4b8eeb0d0347d48b7ee94c7a5334f9bb@192.168.1.84:5060 (Checking To) --From tag as46033be3 --To-tag [Aug 10 15:52:24] DEBUG[17897] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #385 [Aug 10 15:52:24] DEBUG[17897] chan_sip.c: Stopping retransmission on '4b8eeb0d0347d48b7ee94c7a5334f9bb@192.168.1.84:5060' of Request 102: Match Found [Aug 10 15:52:24] DEBUG[17897] chan_sip.c: Destroying SIP dialog 4b8eeb0d0347d48b7ee94c7a5334f9bb@192.168.1.84:5060 [Aug 10 15:52:24] VERBOSE[17897] chan_sip.c: Really destroying SIP dialog '4b8eeb0d0347d48b7ee94c7a5334f9bb@192.168.1.84:5060' Method: OPTIONS [Aug 10 15:52:24] DEBUG[17897] chan_sip.c: Allocating new SIP dialog for 576ba3401c31e994556bc99462559ecc@127.0.0.1:5060 - OPTIONS (No RTP) [Aug 10 15:52:24] DEBUG[17897] acl.c: For destination '192.168.1.102', our source address is '192.168.1.84'. [Aug 10 15:52:24] DEBUG[17897] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.1.84:5060 [Aug 10 15:52:24] DEBUG[17897] chan_sip.c: Initializing initreq for method OPTIONS - callid 7ffc7bac600005be1d4f9e89196758f1@192.168.1.84:5060 [Aug 10 15:52:24] DEBUG[17897] chan_sip.c: Header 0 [ 57]: OPTIONS sip:2212@192.168.1.102:2048;line=z1zv7jsj SIP/2.0 [Aug 10 15:52:24] DEBUG[17897] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK23ddfcc6;rport [Aug 10 15:52:24] DEBUG[17897] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 10 15:52:24] DEBUG[17897] chan_sip.c: Header 3 [ 59]: From: "asterisk" ;tag=as56bb5924 [Aug 10 15:52:24] DEBUG[17897] chan_sip.c: Header 4 [ 47]: To: [Aug 10 15:52:24] DEBUG[17897] chan_sip.c: Header 5 [ 41]: Contact: [Aug 10 15:52:24] DEBUG[17897] chan_sip.c: Header 6 [ 59]: Call-ID: 7ffc7bac600005be1d4f9e89196758f1@192.168.1.84:5060 [Aug 10 15:52:24] DEBUG[17897] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:52:24] DEBUG[17897] chan_sip.c: Header 8 [ 33]: User-Agent: Asterisk PBX 1.8.15.0 [Aug 10 15:52:24] DEBUG[17897] chan_sip.c: Header 9 [ 35]: Date: Fri, 10 Aug 2012 13:52:24 GMT [Aug 10 15:52:24] DEBUG[17897] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 10 15:52:24] DEBUG[17897] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 10 15:52:24] VERBOSE[17897] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.102:2048: OPTIONS sip:2212@192.168.1.102:2048;line=z1zv7jsj SIP/2.0 Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK23ddfcc6;rport Max-Forwards: 70 From: "asterisk" ;tag=as56bb5924 To: Contact: Call-ID: 7ffc7bac600005be1d4f9e89196758f1@192.168.1.84:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.15.0 Date: Fri, 10 Aug 2012 13:52:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Aug 10 15:52:24] DEBUG[17897] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #388 [Aug 10 15:52:24] DEBUG[17897] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.1.102:2048 [Aug 10 15:52:24] VERBOSE[17897] chan_sip.c: <--- SIP read from UDP:192.168.1.102:2048 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK23ddfcc6;rport=5060 From: "asterisk" ;tag=as56bb5924 To: Call-ID: 7ffc7bac600005be1d4f9e89196758f1@192.168.1.84:5060 CSeq: 102 OPTIONS Contact: ;reg-id=1 User-Agent: snom300/8.4.32 Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Content-Length: 0 <-------------> [Aug 10 15:52:24] DEBUG[17897] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 10 15:52:24] DEBUG[17897] chan_sip.c: Header 1 [ 68]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK23ddfcc6;rport=5060 [Aug 10 15:52:24] DEBUG[17897] chan_sip.c: Header 2 [ 59]: From: "asterisk" ;tag=as56bb5924 [Aug 10 15:52:24] DEBUG[17897] chan_sip.c: Header 3 [ 47]: To: [Aug 10 15:52:24] DEBUG[17897] chan_sip.c: Header 4 [ 59]: Call-ID: 7ffc7bac600005be1d4f9e89196758f1@192.168.1.84:5060 [Aug 10 15:52:24] DEBUG[17897] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:52:24] DEBUG[17897] chan_sip.c: Header 6 [ 61]: Contact: ;reg-id=1 [Aug 10 15:52:24] DEBUG[17897] chan_sip.c: Header 7 [ 26]: User-Agent: snom300/8.4.32 [Aug 10 15:52:24] DEBUG[17897] chan_sip.c: Header 8 [ 19]: Accept-Language: en [Aug 10 15:52:24] DEBUG[17897] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [Aug 10 15:52:24] DEBUG[17897] chan_sip.c: Header 10 [ 96]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE [Aug 10 15:52:24] DEBUG[17897] chan_sip.c: Header 11 [ 42]: Allow-Events: talk, hold, refer, call-info [Aug 10 15:52:24] DEBUG[17897] chan_sip.c: Header 12 [ 47]: Supported: timer, 100rel, replaces, from-change [Aug 10 15:52:24] DEBUG[17897] chan_sip.c: Header 13 [ 17]: Content-Length: 0 [Aug 10 15:52:24] VERBOSE[17897] chan_sip.c: --- (14 headers 0 lines) --- [Aug 10 15:52:24] DEBUG[17897] chan_sip.c: = Looking for Call ID: 7ffc7bac600005be1d4f9e89196758f1@192.168.1.84:5060 (Checking To) --From tag as56bb5924 --To-tag [Aug 10 15:52:24] DEBUG[17897] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #388 [Aug 10 15:52:24] DEBUG[17897] chan_sip.c: Stopping retransmission on '7ffc7bac600005be1d4f9e89196758f1@192.168.1.84:5060' of Request 102: Match Found [Aug 10 15:52:24] DEBUG[17897] chan_sip.c: Destroying SIP dialog 7ffc7bac600005be1d4f9e89196758f1@192.168.1.84:5060 [Aug 10 15:52:24] VERBOSE[17897] chan_sip.c: Really destroying SIP dialog '7ffc7bac600005be1d4f9e89196758f1@192.168.1.84:5060' Method: OPTIONS [Aug 10 15:52:24] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:52:24] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:52:29] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:52:29] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:52:34] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:52:34] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:52:39] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:52:39] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:52:44] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:52:44] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: Allocating new SIP dialog for 1611661334455eec565b66e119eecd39@127.0.0.1:5060 - OPTIONS (No RTP) [Aug 10 15:52:46] DEBUG[17897] acl.c: For destination '192.168.1.106', our source address is '192.168.1.84'. [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.1.84:5060 [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: Initializing initreq for method OPTIONS - callid 309edab44f77260b1131aa4f2bf1c82f@192.168.1.84:5060 [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: Header 0 [ 57]: OPTIONS sip:2219@192.168.1.106:2048;line=7g8ggsvj SIP/2.0 [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK32b6884c;rport [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: Header 3 [ 59]: From: "asterisk" ;tag=as30d7791a [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: Header 4 [ 47]: To: [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: Header 5 [ 41]: Contact: [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: Header 6 [ 59]: Call-ID: 309edab44f77260b1131aa4f2bf1c82f@192.168.1.84:5060 [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: Header 8 [ 33]: User-Agent: Asterisk PBX 1.8.15.0 [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: Header 9 [ 35]: Date: Fri, 10 Aug 2012 13:52:46 GMT [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 10 15:52:46] VERBOSE[17897] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.106:2048: OPTIONS sip:2219@192.168.1.106:2048;line=7g8ggsvj SIP/2.0 Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK32b6884c;rport Max-Forwards: 70 From: "asterisk" ;tag=as30d7791a To: Contact: Call-ID: 309edab44f77260b1131aa4f2bf1c82f@192.168.1.84:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.15.0 Date: Fri, 10 Aug 2012 13:52:46 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #391 [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.1.106:2048 [Aug 10 15:52:46] VERBOSE[17897] chan_sip.c: <--- SIP read from UDP:192.168.1.106:2048 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK32b6884c;rport=5060 From: "asterisk" ;tag=as30d7791a To: ;tag=va9gz102as Call-ID: 309edab44f77260b1131aa4f2bf1c82f@192.168.1.84:5060 CSeq: 102 OPTIONS Contact: ;reg-id=1 User-Agent: snom300/8.7.3.10 Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: 100rel, replaces, from-change Content-Length: 0 <-------------> [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: Header 1 [ 68]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK32b6884c;rport=5060 [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: Header 2 [ 59]: From: "asterisk" ;tag=as30d7791a [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: Header 3 [ 62]: To: ;tag=va9gz102as [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: Header 4 [ 59]: Call-ID: 309edab44f77260b1131aa4f2bf1c82f@192.168.1.84:5060 [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: Header 6 [ 61]: Contact: ;reg-id=1 [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: Header 7 [ 28]: User-Agent: snom300/8.7.3.10 [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: Header 8 [ 19]: Accept-Language: en [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: Header 10 [ 96]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: Header 11 [ 42]: Allow-Events: talk, hold, refer, call-info [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: Header 12 [ 40]: Supported: 100rel, replaces, from-change [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: Header 13 [ 17]: Content-Length: 0 [Aug 10 15:52:46] VERBOSE[17897] chan_sip.c: --- (14 headers 0 lines) --- [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: = Looking for Call ID: 309edab44f77260b1131aa4f2bf1c82f@192.168.1.84:5060 (Checking To) --From tag as30d7791a --To-tag va9gz102as [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #391 [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: Stopping retransmission on '309edab44f77260b1131aa4f2bf1c82f@192.168.1.84:5060' of Request 102: Match Found [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: Destroying SIP dialog 309edab44f77260b1131aa4f2bf1c82f@192.168.1.84:5060 [Aug 10 15:52:46] VERBOSE[17897] chan_sip.c: Really destroying SIP dialog '309edab44f77260b1131aa4f2bf1c82f@192.168.1.84:5060' Method: OPTIONS [Aug 10 15:52:46] VERBOSE[17897] chan_sip.c: <--- SIP read from UDP:192.168.1.106:2048 ---> REGISTER sip:192.168.1.84 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.106:2048;branch=z9hG4bK-njeovdmcnijy;rport From: "2219" ;tag=yrkxlnzhwq To: "2219" Call-ID: 50250e9b9684-8d3g9c7o4hy2 CSeq: 37 REGISTER Max-Forwards: 70 Contact: ;reg-id=1;q=1.0;+sip.instance="";audio;mobility="fixed";duplex="full";description="snom300";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO" User-Agent: snom300/8.7.3.10 Allow-Events: dialog X-Real-IP: 192.168.1.106 Supported: path, gruu Content-Length: 0 <-------------> [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: Header 0 [ 33]: REGISTER sip:192.168.1.84 SIP/2.0 [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: Header 1 [ 69]: Via: SIP/2.0/UDP 192.168.1.106:2048;branch=z9hG4bK-njeovdmcnijy;rport [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: Header 2 [ 51]: From: "2219" ;tag=yrkxlnzhwq [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: Header 3 [ 34]: To: "2219" [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: Header 4 [ 34]: Call-ID: 50250e9b9684-8d3g9c7o4hy2 [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: Header 5 [ 17]: CSeq: 37 REGISTER [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: Header 7 [306]: Contact: ;reg-id=1;q=1.0;+sip.instance="";audio;mobility="fixed";duplex="full";description="snom300";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO" [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: Header 8 [ 28]: User-Agent: snom300/8.7.3.10 [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: Header 9 [ 20]: Allow-Events: dialog [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: Header 10 [ 24]: X-Real-IP: 192.168.1.106 [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: Header 11 [ 21]: Supported: path, gruu [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Aug 10 15:52:46] VERBOSE[17897] chan_sip.c: --- (13 headers 0 lines) --- [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: = Looking for Call ID: 50250e9b9684-8d3g9c7o4hy2 (Checking From) --From tag yrkxlnzhwq --To-tag [Aug 10 15:52:46] DEBUG[17897] acl.c: For destination '192.168.1.106', our source address is '192.168.1.84'. [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.1.84:5060 [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: Allocating new SIP dialog for 50250e9b9684-8d3g9c7o4hy2 - REGISTER (No RTP) [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: Initializing initreq for method REGISTER - callid 50250e9b9684-8d3g9c7o4hy2 [Aug 10 15:52:46] DEBUG[17897] netsock2.c: Splitting '192.168.1.106:2048' into... [Aug 10 15:52:46] DEBUG[17897] netsock2.c: ...host '192.168.1.106' and port '2048'. [Aug 10 15:52:46] VERBOSE[17897] chan_sip.c: Sending to 192.168.1.106:2048 (NAT) [Aug 10 15:52:46] DEBUG[17897] netsock2.c: Splitting '192.168.1.84' into... [Aug 10 15:52:46] DEBUG[17897] netsock2.c: ...host '192.168.1.84' and port ''. [Aug 10 15:52:46] VERBOSE[17897] chan_sip.c: <--- Transmitting (NAT) to 192.168.1.106:2048 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.106:2048;branch=z9hG4bK-njeovdmcnijy;received=192.168.1.106;rport=2048 From: "2219" ;tag=yrkxlnzhwq To: "2219" ;tag=as7f53ec9c Call-ID: 50250e9b9684-8d3g9c7o4hy2 CSeq: 37 REGISTER Server: Asterisk PBX 1.8.15.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="29db84d8" Content-Length: 0 <------------> [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 192.168.1.106:2048 [Aug 10 15:52:46] VERBOSE[17897] chan_sip.c: Scheduling destruction of SIP dialog '50250e9b9684-8d3g9c7o4hy2' in 32000 ms (Method: REGISTER) [Aug 10 15:52:46] VERBOSE[17897] chan_sip.c: <--- SIP read from UDP:192.168.1.106:2048 ---> REGISTER sip:192.168.1.84 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.106:2048;branch=z9hG4bK-yaxvc4ybsuoy;rport From: "2219" ;tag=yrkxlnzhwq To: "2219" Call-ID: 50250e9b9684-8d3g9c7o4hy2 CSeq: 38 REGISTER Max-Forwards: 70 Contact: ;reg-id=1;q=1.0;+sip.instance="";audio;mobility="fixed";duplex="full";description="snom300";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO" User-Agent: snom300/8.7.3.10 Allow-Events: dialog X-Real-IP: 192.168.1.106 Supported: path, gruu Authorization: Digest username="2219",realm="asterisk",nonce="29db84d8",uri="sip:192.168.1.84",response="ffe29111827589dc8b208e9e9af4b33c",algorithm=MD5 Content-Length: 0 <-------------> [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: Header 0 [ 33]: REGISTER sip:192.168.1.84 SIP/2.0 [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: Header 1 [ 69]: Via: SIP/2.0/UDP 192.168.1.106:2048;branch=z9hG4bK-yaxvc4ybsuoy;rport [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: Header 2 [ 51]: From: "2219" ;tag=yrkxlnzhwq [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: Header 3 [ 34]: To: "2219" [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: Header 4 [ 34]: Call-ID: 50250e9b9684-8d3g9c7o4hy2 [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: Header 5 [ 17]: CSeq: 38 REGISTER [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: Header 7 [306]: Contact: ;reg-id=1;q=1.0;+sip.instance="";audio;mobility="fixed";duplex="full";description="snom300";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO" [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: Header 8 [ 28]: User-Agent: snom300/8.7.3.10 [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: Header 9 [ 20]: Allow-Events: dialog [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: Header 10 [ 24]: X-Real-IP: 192.168.1.106 [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: Header 11 [ 21]: Supported: path, gruu [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: Header 12 [152]: Authorization: Digest username="2219",realm="asterisk",nonce="29db84d8",uri="sip:192.168.1.84",response="ffe29111827589dc8b208e9e9af4b33c",algorithm=MD5 [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: Header 13 [ 17]: Content-Length: 0 [Aug 10 15:52:46] VERBOSE[17897] chan_sip.c: --- (14 headers 0 lines) --- [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: = Looking for Call ID: 50250e9b9684-8d3g9c7o4hy2 (Checking From) --From tag yrkxlnzhwq --To-tag [Aug 10 15:52:46] DEBUG[17897] netsock2.c: Splitting '192.168.1.84' into... [Aug 10 15:52:46] DEBUG[17897] netsock2.c: ...host '192.168.1.84' and port ''. [Aug 10 15:52:46] DEBUG[17897] netsock2.c: Splitting '192.168.1.84' into... [Aug 10 15:52:46] DEBUG[17897] netsock2.c: ...host '192.168.1.84' and port ''. [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: Initializing initreq for method REGISTER - callid 50250e9b9684-8d3g9c7o4hy2 [Aug 10 15:52:46] DEBUG[17897] netsock2.c: Splitting '192.168.1.106:2048' into... [Aug 10 15:52:46] DEBUG[17897] netsock2.c: ...host '192.168.1.106' and port '2048'. [Aug 10 15:52:46] VERBOSE[17897] chan_sip.c: Sending to 192.168.1.106:2048 (NAT) [Aug 10 15:52:46] DEBUG[17897] netsock2.c: Splitting '192.168.1.84' into... [Aug 10 15:52:46] DEBUG[17897] netsock2.c: ...host '192.168.1.84' and port ''. [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: Store REGISTER's src-IP:port for call routing. [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: Allocating new SIP dialog for 2fc6be39622ab36c49388fd403aafd22@127.0.0.1:5060 - OPTIONS (No RTP) [Aug 10 15:52:46] DEBUG[17897] acl.c: For destination '192.168.1.106', our source address is '192.168.1.84'. [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.1.84:5060 [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: Initializing initreq for method OPTIONS - callid 4b9c955266ab0d79619a72014130aed8@192.168.1.84:5060 [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: Header 0 [ 57]: OPTIONS sip:2219@192.168.1.106:2048;line=7g8ggsvj SIP/2.0 [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK1fafcfdd;rport [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: Header 3 [ 59]: From: "asterisk" ;tag=as5a1b93bf [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: Header 4 [ 47]: To: [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: Header 5 [ 41]: Contact: [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: Header 6 [ 59]: Call-ID: 4b9c955266ab0d79619a72014130aed8@192.168.1.84:5060 [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: Header 8 [ 33]: User-Agent: Asterisk PBX 1.8.15.0 [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: Header 9 [ 35]: Date: Fri, 10 Aug 2012 13:52:46 GMT [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 10 15:52:46] VERBOSE[17897] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.106:2048: OPTIONS sip:2219@192.168.1.106:2048;line=7g8ggsvj SIP/2.0 Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK1fafcfdd;rport Max-Forwards: 70 From: "asterisk" ;tag=as5a1b93bf To: Contact: Call-ID: 4b9c955266ab0d79619a72014130aed8@192.168.1.84:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.15.0 Date: Fri, 10 Aug 2012 13:52:46 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #396 [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.1.106:2048 [Aug 10 15:52:46] VERBOSE[17897] chan_sip.c: <--- Transmitting (NAT) to 192.168.1.106:2048 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.106:2048;branch=z9hG4bK-yaxvc4ybsuoy;received=192.168.1.106;rport=2048 From: "2219" ;tag=yrkxlnzhwq To: "2219" ;tag=as7f53ec9c Call-ID: 50250e9b9684-8d3g9c7o4hy2 CSeq: 38 REGISTER Server: Asterisk PBX 1.8.15.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Expires: 120 Contact: ;expires=120 Date: Fri, 10 Aug 2012 13:52:46 GMT Content-Length: 0 <------------> [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.1.106:2048 [Aug 10 15:52:46] VERBOSE[17897] chan_sip.c: Scheduling destruction of SIP dialog '50250e9b9684-8d3g9c7o4hy2' in 32000 ms (Method: REGISTER) [Aug 10 15:52:46] DEBUG[17874] devicestate.c: No provider found, checking channel drivers for SIP - 2219 [Aug 10 15:52:46] DEBUG[17874] chan_sip.c: Checking device state for peer 2219 [Aug 10 15:52:46] DEBUG[17874] devicestate.c: Changing state for SIP/2219 - state 1 (Not in use) [Aug 10 15:52:46] DEBUG[17874] devicestate.c: device 'SIP/2219' state '1' [Aug 10 15:52:46] DEBUG[17907] app_queue.c: Device 'SIP/2219' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Aug 10 15:52:46] VERBOSE[17897] chan_sip.c: <--- SIP read from UDP:192.168.1.106:2048 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK1fafcfdd;rport=5060 From: "asterisk" ;tag=as5a1b93bf To: ;tag=qzst9slzrf Call-ID: 4b9c955266ab0d79619a72014130aed8@192.168.1.84:5060 CSeq: 102 OPTIONS Contact: ;reg-id=1 User-Agent: snom300/8.7.3.10 Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: 100rel, replaces, from-change Content-Length: 0 <-------------> [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: Header 1 [ 68]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK1fafcfdd;rport=5060 [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: Header 2 [ 59]: From: "asterisk" ;tag=as5a1b93bf [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: Header 3 [ 62]: To: ;tag=qzst9slzrf [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: Header 4 [ 59]: Call-ID: 4b9c955266ab0d79619a72014130aed8@192.168.1.84:5060 [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: Header 6 [ 61]: Contact: ;reg-id=1 [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: Header 7 [ 28]: User-Agent: snom300/8.7.3.10 [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: Header 8 [ 19]: Accept-Language: en [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: Header 10 [ 96]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: Header 11 [ 42]: Allow-Events: talk, hold, refer, call-info [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: Header 12 [ 40]: Supported: 100rel, replaces, from-change [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: Header 13 [ 17]: Content-Length: 0 [Aug 10 15:52:46] VERBOSE[17897] chan_sip.c: --- (14 headers 0 lines) --- [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: = Looking for Call ID: 4b9c955266ab0d79619a72014130aed8@192.168.1.84:5060 (Checking To) --From tag as5a1b93bf --To-tag qzst9slzrf [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #396 [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: Stopping retransmission on '4b9c955266ab0d79619a72014130aed8@192.168.1.84:5060' of Request 102: Match Found [Aug 10 15:52:46] DEBUG[17897] chan_sip.c: Destroying SIP dialog 4b9c955266ab0d79619a72014130aed8@192.168.1.84:5060 [Aug 10 15:52:46] VERBOSE[17897] chan_sip.c: Really destroying SIP dialog '4b9c955266ab0d79619a72014130aed8@192.168.1.84:5060' Method: OPTIONS [Aug 10 15:52:49] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:52:49] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:52:54] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:52:54] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:52:59] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:52:59] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:53:04] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:53:04] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:53:09] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:53:09] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:53:14] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:53:14] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:53:18] DEBUG[17897] chan_sip.c: Auto destroying SIP dialog '50250e9b9684-8d3g9c7o4hy2' [Aug 10 15:53:18] DEBUG[17897] chan_sip.c: Destroying SIP dialog 50250e9b9684-8d3g9c7o4hy2 [Aug 10 15:53:18] VERBOSE[17897] chan_sip.c: Really destroying SIP dialog '50250e9b9684-8d3g9c7o4hy2' Method: REGISTER [Aug 10 15:53:19] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:53:19] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:53:24] DEBUG[17897] chan_sip.c: Allocating new SIP dialog for 698471eb6fe235496db71b0151dcbd5f@127.0.0.1:5060 - OPTIONS (No RTP) [Aug 10 15:53:24] DEBUG[17897] acl.c: For destination '192.168.2.210', our source address is '192.168.1.84'. [Aug 10 15:53:24] DEBUG[17897] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.1.84:5060 [Aug 10 15:53:24] DEBUG[17897] chan_sip.c: Initializing initreq for method OPTIONS - callid 79e16769685fe3e42f009c9654b7c94f@192.168.1.84:5060 [Aug 10 15:53:24] DEBUG[17897] chan_sip.c: Header 0 [ 43]: OPTIONS sip:2210@192.168.2.210:2048 SIP/2.0 [Aug 10 15:53:24] DEBUG[17897] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK30be3d97;rport [Aug 10 15:53:24] DEBUG[17897] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 10 15:53:24] DEBUG[17897] chan_sip.c: Header 3 [ 59]: From: "asterisk" ;tag=as4e6ec7e0 [Aug 10 15:53:24] DEBUG[17897] chan_sip.c: Header 4 [ 33]: To: [Aug 10 15:53:24] DEBUG[17897] chan_sip.c: Header 5 [ 41]: Contact: [Aug 10 15:53:24] DEBUG[17897] chan_sip.c: Header 6 [ 59]: Call-ID: 79e16769685fe3e42f009c9654b7c94f@192.168.1.84:5060 [Aug 10 15:53:24] DEBUG[17897] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:53:24] DEBUG[17897] chan_sip.c: Header 8 [ 33]: User-Agent: Asterisk PBX 1.8.15.0 [Aug 10 15:53:24] DEBUG[17897] chan_sip.c: Header 9 [ 35]: Date: Fri, 10 Aug 2012 13:53:24 GMT [Aug 10 15:53:24] DEBUG[17897] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 10 15:53:24] DEBUG[17897] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 10 15:53:24] VERBOSE[17897] chan_sip.c: Reliably Transmitting (NAT) to 192.168.2.210:2048: OPTIONS sip:2210@192.168.2.210:2048 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK30be3d97;rport Max-Forwards: 70 From: "asterisk" ;tag=as4e6ec7e0 To: Contact: Call-ID: 79e16769685fe3e42f009c9654b7c94f@192.168.1.84:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.15.0 Date: Fri, 10 Aug 2012 13:53:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Aug 10 15:53:24] DEBUG[17897] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #400 [Aug 10 15:53:24] DEBUG[17897] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.2.210:2048 [Aug 10 15:53:24] VERBOSE[17897] chan_sip.c: <--- SIP read from UDP:192.168.2.210:2048 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK30be3d97;rport=5060 From: "asterisk" ;tag=as4e6ec7e0 To: Call-ID: 79e16769685fe3e42f009c9654b7c94f@192.168.1.84:5060 CSeq: 102 OPTIONS Contact: ;reg-id=1 User-Agent: snom300/8.4.32 Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: 100rel, replaces, from-change Content-Length: 0 <-------------> [Aug 10 15:53:24] DEBUG[17897] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 10 15:53:24] DEBUG[17897] chan_sip.c: Header 1 [ 68]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK30be3d97;rport=5060 [Aug 10 15:53:24] DEBUG[17897] chan_sip.c: Header 2 [ 59]: From: "asterisk" ;tag=as4e6ec7e0 [Aug 10 15:53:24] DEBUG[17897] chan_sip.c: Header 3 [ 33]: To: [Aug 10 15:53:24] DEBUG[17897] chan_sip.c: Header 4 [ 59]: Call-ID: 79e16769685fe3e42f009c9654b7c94f@192.168.1.84:5060 [Aug 10 15:53:24] DEBUG[17897] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:53:24] DEBUG[17897] chan_sip.c: Header 6 [ 47]: Contact: ;reg-id=1 [Aug 10 15:53:24] DEBUG[17897] chan_sip.c: Header 7 [ 26]: User-Agent: snom300/8.4.32 [Aug 10 15:53:24] DEBUG[17897] chan_sip.c: Header 8 [ 19]: Accept-Language: en [Aug 10 15:53:24] DEBUG[17897] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [Aug 10 15:53:24] DEBUG[17897] chan_sip.c: Header 10 [ 96]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE [Aug 10 15:53:24] DEBUG[17897] chan_sip.c: Header 11 [ 42]: Allow-Events: talk, hold, refer, call-info [Aug 10 15:53:24] DEBUG[17897] chan_sip.c: Header 12 [ 40]: Supported: 100rel, replaces, from-change [Aug 10 15:53:24] DEBUG[17897] chan_sip.c: Header 13 [ 17]: Content-Length: 0 [Aug 10 15:53:24] VERBOSE[17897] chan_sip.c: --- (14 headers 0 lines) --- [Aug 10 15:53:24] DEBUG[17897] chan_sip.c: = Looking for Call ID: 79e16769685fe3e42f009c9654b7c94f@192.168.1.84:5060 (Checking To) --From tag as4e6ec7e0 --To-tag [Aug 10 15:53:24] DEBUG[17897] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #400 [Aug 10 15:53:24] DEBUG[17897] chan_sip.c: Stopping retransmission on '79e16769685fe3e42f009c9654b7c94f@192.168.1.84:5060' of Request 102: Match Found [Aug 10 15:53:24] DEBUG[17897] chan_sip.c: Destroying SIP dialog 79e16769685fe3e42f009c9654b7c94f@192.168.1.84:5060 [Aug 10 15:53:24] VERBOSE[17897] chan_sip.c: Really destroying SIP dialog '79e16769685fe3e42f009c9654b7c94f@192.168.1.84:5060' Method: OPTIONS [Aug 10 15:53:24] DEBUG[17897] chan_sip.c: Allocating new SIP dialog for 57163d6d047ec16251ddad3910c08442@127.0.0.1:5060 - OPTIONS (No RTP) [Aug 10 15:53:24] DEBUG[17897] acl.c: For destination '192.168.1.102', our source address is '192.168.1.84'. [Aug 10 15:53:24] DEBUG[17897] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.1.84:5060 [Aug 10 15:53:24] DEBUG[17897] chan_sip.c: Initializing initreq for method OPTIONS - callid 3847df73721c72795fa22298473e4618@192.168.1.84:5060 [Aug 10 15:53:24] DEBUG[17897] chan_sip.c: Header 0 [ 57]: OPTIONS sip:2212@192.168.1.102:2048;line=z1zv7jsj SIP/2.0 [Aug 10 15:53:24] DEBUG[17897] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK6ea1df4f;rport [Aug 10 15:53:24] DEBUG[17897] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 10 15:53:24] DEBUG[17897] chan_sip.c: Header 3 [ 59]: From: "asterisk" ;tag=as6399a6d6 [Aug 10 15:53:24] DEBUG[17897] chan_sip.c: Header 4 [ 47]: To: [Aug 10 15:53:24] DEBUG[17897] chan_sip.c: Header 5 [ 41]: Contact: [Aug 10 15:53:24] DEBUG[17897] chan_sip.c: Header 6 [ 59]: Call-ID: 3847df73721c72795fa22298473e4618@192.168.1.84:5060 [Aug 10 15:53:24] DEBUG[17897] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:53:24] DEBUG[17897] chan_sip.c: Header 8 [ 33]: User-Agent: Asterisk PBX 1.8.15.0 [Aug 10 15:53:24] DEBUG[17897] chan_sip.c: Header 9 [ 35]: Date: Fri, 10 Aug 2012 13:53:24 GMT [Aug 10 15:53:24] DEBUG[17897] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 10 15:53:24] DEBUG[17897] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 10 15:53:24] VERBOSE[17897] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.102:2048: OPTIONS sip:2212@192.168.1.102:2048;line=z1zv7jsj SIP/2.0 Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK6ea1df4f;rport Max-Forwards: 70 From: "asterisk" ;tag=as6399a6d6 To: Contact: Call-ID: 3847df73721c72795fa22298473e4618@192.168.1.84:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.15.0 Date: Fri, 10 Aug 2012 13:53:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Aug 10 15:53:24] DEBUG[17897] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #403 [Aug 10 15:53:24] DEBUG[17897] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.1.102:2048 [Aug 10 15:53:24] VERBOSE[17897] chan_sip.c: <--- SIP read from UDP:192.168.1.102:2048 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK6ea1df4f;rport=5060 From: "asterisk" ;tag=as6399a6d6 To: Call-ID: 3847df73721c72795fa22298473e4618@192.168.1.84:5060 CSeq: 102 OPTIONS Contact: ;reg-id=1 User-Agent: snom300/8.4.32 Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Content-Length: 0 <-------------> [Aug 10 15:53:24] DEBUG[17897] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 10 15:53:24] DEBUG[17897] chan_sip.c: Header 1 [ 68]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK6ea1df4f;rport=5060 [Aug 10 15:53:24] DEBUG[17897] chan_sip.c: Header 2 [ 59]: From: "asterisk" ;tag=as6399a6d6 [Aug 10 15:53:24] DEBUG[17897] chan_sip.c: Header 3 [ 47]: To: [Aug 10 15:53:24] DEBUG[17897] chan_sip.c: Header 4 [ 59]: Call-ID: 3847df73721c72795fa22298473e4618@192.168.1.84:5060 [Aug 10 15:53:24] DEBUG[17897] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:53:24] DEBUG[17897] chan_sip.c: Header 6 [ 61]: Contact: ;reg-id=1 [Aug 10 15:53:24] DEBUG[17897] chan_sip.c: Header 7 [ 26]: User-Agent: snom300/8.4.32 [Aug 10 15:53:24] DEBUG[17897] chan_sip.c: Header 8 [ 19]: Accept-Language: en [Aug 10 15:53:24] DEBUG[17897] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [Aug 10 15:53:24] DEBUG[17897] chan_sip.c: Header 10 [ 96]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE [Aug 10 15:53:24] DEBUG[17897] chan_sip.c: Header 11 [ 42]: Allow-Events: talk, hold, refer, call-info [Aug 10 15:53:24] DEBUG[17897] chan_sip.c: Header 12 [ 47]: Supported: timer, 100rel, replaces, from-change [Aug 10 15:53:24] DEBUG[17897] chan_sip.c: Header 13 [ 17]: Content-Length: 0 [Aug 10 15:53:24] VERBOSE[17897] chan_sip.c: --- (14 headers 0 lines) --- [Aug 10 15:53:24] DEBUG[17897] chan_sip.c: = Looking for Call ID: 3847df73721c72795fa22298473e4618@192.168.1.84:5060 (Checking To) --From tag as6399a6d6 --To-tag [Aug 10 15:53:24] DEBUG[17897] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #403 [Aug 10 15:53:24] DEBUG[17897] chan_sip.c: Stopping retransmission on '3847df73721c72795fa22298473e4618@192.168.1.84:5060' of Request 102: Match Found [Aug 10 15:53:24] DEBUG[17897] chan_sip.c: Destroying SIP dialog 3847df73721c72795fa22298473e4618@192.168.1.84:5060 [Aug 10 15:53:24] VERBOSE[17897] chan_sip.c: Really destroying SIP dialog '3847df73721c72795fa22298473e4618@192.168.1.84:5060' Method: OPTIONS [Aug 10 15:53:24] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:53:24] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:53:29] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:53:29] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:53:34] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:53:34] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:53:39] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:53:39] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:53:44] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:53:44] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: Allocating new SIP dialog for 36ebc13e3d8ff8c4260886082ff17bd5@127.0.0.1:5060 - OPTIONS (No RTP) [Aug 10 15:53:46] DEBUG[17897] acl.c: For destination '192.168.1.106', our source address is '192.168.1.84'. [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.1.84:5060 [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: Initializing initreq for method OPTIONS - callid 089583aa6e9aa85c5114c48d0701ddb8@192.168.1.84:5060 [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: Header 0 [ 57]: OPTIONS sip:2219@192.168.1.106:2048;line=7g8ggsvj SIP/2.0 [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK6f7deeb9;rport [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: Header 3 [ 59]: From: "asterisk" ;tag=as4efd1bd8 [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: Header 4 [ 47]: To: [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: Header 5 [ 41]: Contact: [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: Header 6 [ 59]: Call-ID: 089583aa6e9aa85c5114c48d0701ddb8@192.168.1.84:5060 [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: Header 8 [ 33]: User-Agent: Asterisk PBX 1.8.15.0 [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: Header 9 [ 35]: Date: Fri, 10 Aug 2012 13:53:46 GMT [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 10 15:53:46] VERBOSE[17897] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.106:2048: OPTIONS sip:2219@192.168.1.106:2048;line=7g8ggsvj SIP/2.0 Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK6f7deeb9;rport Max-Forwards: 70 From: "asterisk" ;tag=as4efd1bd8 To: Contact: Call-ID: 089583aa6e9aa85c5114c48d0701ddb8@192.168.1.84:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.15.0 Date: Fri, 10 Aug 2012 13:53:46 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #406 [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.1.106:2048 [Aug 10 15:53:46] VERBOSE[17897] chan_sip.c: <--- SIP read from UDP:192.168.1.106:2048 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK6f7deeb9;rport=5060 From: "asterisk" ;tag=as4efd1bd8 To: ;tag=0zswgek40k Call-ID: 089583aa6e9aa85c5114c48d0701ddb8@192.168.1.84:5060 CSeq: 102 OPTIONS Contact: ;reg-id=1 User-Agent: snom300/8.7.3.10 Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: 100rel, replaces, from-change Content-Length: 0 <-------------> [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: Header 1 [ 68]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK6f7deeb9;rport=5060 [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: Header 2 [ 59]: From: "asterisk" ;tag=as4efd1bd8 [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: Header 3 [ 62]: To: ;tag=0zswgek40k [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: Header 4 [ 59]: Call-ID: 089583aa6e9aa85c5114c48d0701ddb8@192.168.1.84:5060 [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: Header 6 [ 61]: Contact: ;reg-id=1 [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: Header 7 [ 28]: User-Agent: snom300/8.7.3.10 [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: Header 8 [ 19]: Accept-Language: en [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: Header 10 [ 96]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: Header 11 [ 42]: Allow-Events: talk, hold, refer, call-info [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: Header 12 [ 40]: Supported: 100rel, replaces, from-change [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: Header 13 [ 17]: Content-Length: 0 [Aug 10 15:53:46] VERBOSE[17897] chan_sip.c: --- (14 headers 0 lines) --- [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: = Looking for Call ID: 089583aa6e9aa85c5114c48d0701ddb8@192.168.1.84:5060 (Checking To) --From tag as4efd1bd8 --To-tag 0zswgek40k [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #406 [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: Stopping retransmission on '089583aa6e9aa85c5114c48d0701ddb8@192.168.1.84:5060' of Request 102: Match Found [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: Destroying SIP dialog 089583aa6e9aa85c5114c48d0701ddb8@192.168.1.84:5060 [Aug 10 15:53:46] VERBOSE[17897] chan_sip.c: Really destroying SIP dialog '089583aa6e9aa85c5114c48d0701ddb8@192.168.1.84:5060' Method: OPTIONS [Aug 10 15:53:46] VERBOSE[17897] chan_sip.c: <--- SIP read from UDP:192.168.1.106:2048 ---> REGISTER sip:192.168.1.84 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.106:2048;branch=z9hG4bK-x3okbviyio3y;rport From: "2219" ;tag=3r0bk9km9r To: "2219" Call-ID: 50250e9b9684-8d3g9c7o4hy2 CSeq: 39 REGISTER Max-Forwards: 70 Contact: ;reg-id=1;q=1.0;+sip.instance="";audio;mobility="fixed";duplex="full";description="snom300";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO" User-Agent: snom300/8.7.3.10 Allow-Events: dialog X-Real-IP: 192.168.1.106 Supported: path, gruu Content-Length: 0 <-------------> [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: Header 0 [ 33]: REGISTER sip:192.168.1.84 SIP/2.0 [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: Header 1 [ 69]: Via: SIP/2.0/UDP 192.168.1.106:2048;branch=z9hG4bK-x3okbviyio3y;rport [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: Header 2 [ 51]: From: "2219" ;tag=3r0bk9km9r [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: Header 3 [ 34]: To: "2219" [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: Header 4 [ 34]: Call-ID: 50250e9b9684-8d3g9c7o4hy2 [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: Header 5 [ 17]: CSeq: 39 REGISTER [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: Header 7 [306]: Contact: ;reg-id=1;q=1.0;+sip.instance="";audio;mobility="fixed";duplex="full";description="snom300";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO" [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: Header 8 [ 28]: User-Agent: snom300/8.7.3.10 [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: Header 9 [ 20]: Allow-Events: dialog [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: Header 10 [ 24]: X-Real-IP: 192.168.1.106 [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: Header 11 [ 21]: Supported: path, gruu [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Aug 10 15:53:46] VERBOSE[17897] chan_sip.c: --- (13 headers 0 lines) --- [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: = Looking for Call ID: 50250e9b9684-8d3g9c7o4hy2 (Checking From) --From tag 3r0bk9km9r --To-tag [Aug 10 15:53:46] DEBUG[17897] acl.c: For destination '192.168.1.106', our source address is '192.168.1.84'. [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.1.84:5060 [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: Allocating new SIP dialog for 50250e9b9684-8d3g9c7o4hy2 - REGISTER (No RTP) [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: Initializing initreq for method REGISTER - callid 50250e9b9684-8d3g9c7o4hy2 [Aug 10 15:53:46] DEBUG[17897] netsock2.c: Splitting '192.168.1.106:2048' into... [Aug 10 15:53:46] DEBUG[17897] netsock2.c: ...host '192.168.1.106' and port '2048'. [Aug 10 15:53:46] VERBOSE[17897] chan_sip.c: Sending to 192.168.1.106:2048 (NAT) [Aug 10 15:53:46] DEBUG[17897] netsock2.c: Splitting '192.168.1.84' into... [Aug 10 15:53:46] DEBUG[17897] netsock2.c: ...host '192.168.1.84' and port ''. [Aug 10 15:53:46] VERBOSE[17897] chan_sip.c: <--- Transmitting (NAT) to 192.168.1.106:2048 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.106:2048;branch=z9hG4bK-x3okbviyio3y;received=192.168.1.106;rport=2048 From: "2219" ;tag=3r0bk9km9r To: "2219" ;tag=as17258fe1 Call-ID: 50250e9b9684-8d3g9c7o4hy2 CSeq: 39 REGISTER Server: Asterisk PBX 1.8.15.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2e72bed6" Content-Length: 0 <------------> [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 192.168.1.106:2048 [Aug 10 15:53:46] VERBOSE[17897] chan_sip.c: Scheduling destruction of SIP dialog '50250e9b9684-8d3g9c7o4hy2' in 32000 ms (Method: REGISTER) [Aug 10 15:53:46] VERBOSE[17897] chan_sip.c: <--- SIP read from UDP:192.168.1.106:2048 ---> REGISTER sip:192.168.1.84 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.106:2048;branch=z9hG4bK-9ykaahxi19d3;rport From: "2219" ;tag=3r0bk9km9r To: "2219" Call-ID: 50250e9b9684-8d3g9c7o4hy2 CSeq: 40 REGISTER Max-Forwards: 70 Contact: ;reg-id=1;q=1.0;+sip.instance="";audio;mobility="fixed";duplex="full";description="snom300";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO" User-Agent: snom300/8.7.3.10 Allow-Events: dialog X-Real-IP: 192.168.1.106 Supported: path, gruu Authorization: Digest username="2219",realm="asterisk",nonce="2e72bed6",uri="sip:192.168.1.84",response="91d4529ba30609a0d75b304a7c42df38",algorithm=MD5 Content-Length: 0 <-------------> [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: Header 0 [ 33]: REGISTER sip:192.168.1.84 SIP/2.0 [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: Header 1 [ 69]: Via: SIP/2.0/UDP 192.168.1.106:2048;branch=z9hG4bK-9ykaahxi19d3;rport [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: Header 2 [ 51]: From: "2219" ;tag=3r0bk9km9r [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: Header 3 [ 34]: To: "2219" [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: Header 4 [ 34]: Call-ID: 50250e9b9684-8d3g9c7o4hy2 [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: Header 5 [ 17]: CSeq: 40 REGISTER [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: Header 7 [306]: Contact: ;reg-id=1;q=1.0;+sip.instance="";audio;mobility="fixed";duplex="full";description="snom300";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO" [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: Header 8 [ 28]: User-Agent: snom300/8.7.3.10 [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: Header 9 [ 20]: Allow-Events: dialog [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: Header 10 [ 24]: X-Real-IP: 192.168.1.106 [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: Header 11 [ 21]: Supported: path, gruu [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: Header 12 [152]: Authorization: Digest username="2219",realm="asterisk",nonce="2e72bed6",uri="sip:192.168.1.84",response="91d4529ba30609a0d75b304a7c42df38",algorithm=MD5 [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: Header 13 [ 17]: Content-Length: 0 [Aug 10 15:53:46] VERBOSE[17897] chan_sip.c: --- (14 headers 0 lines) --- [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: = Looking for Call ID: 50250e9b9684-8d3g9c7o4hy2 (Checking From) --From tag 3r0bk9km9r --To-tag [Aug 10 15:53:46] DEBUG[17897] netsock2.c: Splitting '192.168.1.84' into... [Aug 10 15:53:46] DEBUG[17897] netsock2.c: ...host '192.168.1.84' and port ''. [Aug 10 15:53:46] DEBUG[17897] netsock2.c: Splitting '192.168.1.84' into... [Aug 10 15:53:46] DEBUG[17897] netsock2.c: ...host '192.168.1.84' and port ''. [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: Initializing initreq for method REGISTER - callid 50250e9b9684-8d3g9c7o4hy2 [Aug 10 15:53:46] DEBUG[17897] netsock2.c: Splitting '192.168.1.106:2048' into... [Aug 10 15:53:46] DEBUG[17897] netsock2.c: ...host '192.168.1.106' and port '2048'. [Aug 10 15:53:46] VERBOSE[17897] chan_sip.c: Sending to 192.168.1.106:2048 (NAT) [Aug 10 15:53:46] DEBUG[17897] netsock2.c: Splitting '192.168.1.84' into... [Aug 10 15:53:46] DEBUG[17897] netsock2.c: ...host '192.168.1.84' and port ''. [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: Store REGISTER's src-IP:port for call routing. [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: Allocating new SIP dialog for 6ae8dc5e6bf898bb2e6d58891e6267f6@127.0.0.1:5060 - OPTIONS (No RTP) [Aug 10 15:53:46] DEBUG[17897] acl.c: For destination '192.168.1.106', our source address is '192.168.1.84'. [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.1.84:5060 [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: Initializing initreq for method OPTIONS - callid 11c04c3835c59e921a860fde2df67717@192.168.1.84:5060 [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: Header 0 [ 57]: OPTIONS sip:2219@192.168.1.106:2048;line=7g8ggsvj SIP/2.0 [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK65866ed0;rport [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: Header 3 [ 59]: From: "asterisk" ;tag=as302c5f55 [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: Header 4 [ 47]: To: [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: Header 5 [ 41]: Contact: [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: Header 6 [ 59]: Call-ID: 11c04c3835c59e921a860fde2df67717@192.168.1.84:5060 [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: Header 8 [ 33]: User-Agent: Asterisk PBX 1.8.15.0 [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: Header 9 [ 35]: Date: Fri, 10 Aug 2012 13:53:46 GMT [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 10 15:53:46] VERBOSE[17897] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.106:2048: OPTIONS sip:2219@192.168.1.106:2048;line=7g8ggsvj SIP/2.0 Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK65866ed0;rport Max-Forwards: 70 From: "asterisk" ;tag=as302c5f55 To: Contact: Call-ID: 11c04c3835c59e921a860fde2df67717@192.168.1.84:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.15.0 Date: Fri, 10 Aug 2012 13:53:46 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #411 [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.1.106:2048 [Aug 10 15:53:46] VERBOSE[17897] chan_sip.c: <--- Transmitting (NAT) to 192.168.1.106:2048 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.106:2048;branch=z9hG4bK-9ykaahxi19d3;received=192.168.1.106;rport=2048 From: "2219" ;tag=3r0bk9km9r To: "2219" ;tag=as17258fe1 Call-ID: 50250e9b9684-8d3g9c7o4hy2 CSeq: 40 REGISTER Server: Asterisk PBX 1.8.15.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Expires: 120 Contact: ;expires=120 Date: Fri, 10 Aug 2012 13:53:46 GMT Content-Length: 0 <------------> [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.1.106:2048 [Aug 10 15:53:46] VERBOSE[17897] chan_sip.c: Scheduling destruction of SIP dialog '50250e9b9684-8d3g9c7o4hy2' in 32000 ms (Method: REGISTER) [Aug 10 15:53:46] DEBUG[17874] devicestate.c: No provider found, checking channel drivers for SIP - 2219 [Aug 10 15:53:46] DEBUG[17874] chan_sip.c: Checking device state for peer 2219 [Aug 10 15:53:46] DEBUG[17874] devicestate.c: Changing state for SIP/2219 - state 1 (Not in use) [Aug 10 15:53:46] DEBUG[17874] devicestate.c: device 'SIP/2219' state '1' [Aug 10 15:53:46] DEBUG[17907] app_queue.c: Device 'SIP/2219' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Aug 10 15:53:46] VERBOSE[17897] chan_sip.c: <--- SIP read from UDP:192.168.1.106:2048 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK65866ed0;rport=5060 From: "asterisk" ;tag=as302c5f55 To: ;tag=r77ry2i1eo Call-ID: 11c04c3835c59e921a860fde2df67717@192.168.1.84:5060 CSeq: 102 OPTIONS Contact: ;reg-id=1 User-Agent: snom300/8.7.3.10 Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: 100rel, replaces, from-change Content-Length: 0 <-------------> [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: Header 1 [ 68]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK65866ed0;rport=5060 [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: Header 2 [ 59]: From: "asterisk" ;tag=as302c5f55 [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: Header 3 [ 62]: To: ;tag=r77ry2i1eo [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: Header 4 [ 59]: Call-ID: 11c04c3835c59e921a860fde2df67717@192.168.1.84:5060 [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: Header 6 [ 61]: Contact: ;reg-id=1 [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: Header 7 [ 28]: User-Agent: snom300/8.7.3.10 [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: Header 8 [ 19]: Accept-Language: en [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: Header 10 [ 96]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: Header 11 [ 42]: Allow-Events: talk, hold, refer, call-info [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: Header 12 [ 40]: Supported: 100rel, replaces, from-change [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: Header 13 [ 17]: Content-Length: 0 [Aug 10 15:53:46] VERBOSE[17897] chan_sip.c: --- (14 headers 0 lines) --- [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: = Looking for Call ID: 11c04c3835c59e921a860fde2df67717@192.168.1.84:5060 (Checking To) --From tag as302c5f55 --To-tag r77ry2i1eo [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #411 [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: Stopping retransmission on '11c04c3835c59e921a860fde2df67717@192.168.1.84:5060' of Request 102: Match Found [Aug 10 15:53:46] DEBUG[17897] chan_sip.c: Destroying SIP dialog 11c04c3835c59e921a860fde2df67717@192.168.1.84:5060 [Aug 10 15:53:46] VERBOSE[17897] chan_sip.c: Really destroying SIP dialog '11c04c3835c59e921a860fde2df67717@192.168.1.84:5060' Method: OPTIONS [Aug 10 15:53:49] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:53:49] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:53:54] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:53:54] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:53:59] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:53:59] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:54:04] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:54:04] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:54:09] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:54:09] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:54:14] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:54:14] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:54:18] DEBUG[17897] chan_sip.c: Auto destroying SIP dialog '50250e9b9684-8d3g9c7o4hy2' [Aug 10 15:54:18] DEBUG[17897] chan_sip.c: Destroying SIP dialog 50250e9b9684-8d3g9c7o4hy2 [Aug 10 15:54:18] VERBOSE[17897] chan_sip.c: Really destroying SIP dialog '50250e9b9684-8d3g9c7o4hy2' Method: REGISTER [Aug 10 15:54:19] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:54:19] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:54:21] VERBOSE[17897] chan_sip.c: <--- SIP read from UDP:192.168.2.210:2048 ---> INVITE sip:2212@192.168.1.84:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.210:2048;branch=z9hG4bK-ng8nnwgz15g4;rport From: "2210" ;tag=llqi1mlu1l To: ;tag=as77bb01b2 Call-ID: 3c2672a5cbc0-o7llh150d2xu CSeq: 3 INVITE Max-Forwards: 70 Contact: ;reg-id=1 X-Serialnumber: 0004132500A7 P-Key-Flags: keys="3" User-Agent: snom300/8.4.32 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: 100rel, replaces, from-change Content-Type: application/sdp Content-Length: 349 v=0 o=root 2048061303 2048061304 IN IP4 192.168.2.210 s=call c=IN IP4 192.168.2.210 t=0 0 m=audio 17574 RTP/SAVP 8 0 101 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:i99dpczMNuFrrIJkKr09+4nfcuXTzsz5W11qjb74 a=direction:both a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendonly <-------------> [Aug 10 15:54:21] DEBUG[17897] chan_sip.c: Header 0 [ 41]: INVITE sip:2212@192.168.1.84:5060 SIP/2.0 [Aug 10 15:54:21] DEBUG[17897] chan_sip.c: Header 1 [ 69]: Via: SIP/2.0/UDP 192.168.2.210:2048;branch=z9hG4bK-ng8nnwgz15g4;rport [Aug 10 15:54:21] DEBUG[17897] chan_sip.c: Header 2 [ 51]: From: "2210" ;tag=llqi1mlu1l [Aug 10 15:54:21] DEBUG[17897] chan_sip.c: Header 3 [ 42]: To: ;tag=as77bb01b2 [Aug 10 15:54:21] DEBUG[17897] chan_sip.c: Header 4 [ 34]: Call-ID: 3c2672a5cbc0-o7llh150d2xu [Aug 10 15:54:21] DEBUG[17897] chan_sip.c: Header 5 [ 14]: CSeq: 3 INVITE [Aug 10 15:54:21] DEBUG[17897] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Aug 10 15:54:21] DEBUG[17897] chan_sip.c: Header 7 [ 47]: Contact: ;reg-id=1 [Aug 10 15:54:21] DEBUG[17897] chan_sip.c: Header 8 [ 28]: X-Serialnumber: 0004132500A7 [Aug 10 15:54:21] DEBUG[17897] chan_sip.c: Header 9 [ 21]: P-Key-Flags: keys="3" [Aug 10 15:54:21] DEBUG[17897] chan_sip.c: Header 10 [ 26]: User-Agent: snom300/8.4.32 [Aug 10 15:54:21] DEBUG[17897] chan_sip.c: Header 11 [ 23]: Accept: application/sdp [Aug 10 15:54:21] DEBUG[17897] chan_sip.c: Header 12 [ 96]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE [Aug 10 15:54:21] DEBUG[17897] chan_sip.c: Header 13 [ 42]: Allow-Events: talk, hold, refer, call-info [Aug 10 15:54:21] DEBUG[17897] chan_sip.c: Header 14 [ 40]: Supported: 100rel, replaces, from-change [Aug 10 15:54:21] DEBUG[17897] chan_sip.c: Header 15 [ 29]: Content-Type: application/sdp [Aug 10 15:54:21] DEBUG[17897] chan_sip.c: Header 16 [ 19]: Content-Length: 349 [Aug 10 15:54:21] DEBUG[17897] chan_sip.c: Header 17 [ 0]: [Aug 10 15:54:21] DEBUG[17897] chan_sip.c: Body 0 [ 3]: v=0 [Aug 10 15:54:21] DEBUG[17897] chan_sip.c: Body 1 [ 49]: o=root 2048061303 2048061304 IN IP4 192.168.2.210 [Aug 10 15:54:21] DEBUG[17897] chan_sip.c: Body 2 [ 6]: s=call [Aug 10 15:54:21] DEBUG[17897] chan_sip.c: Body 3 [ 22]: c=IN IP4 192.168.2.210 [Aug 10 15:54:21] DEBUG[17897] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 10 15:54:21] DEBUG[17897] chan_sip.c: Body 5 [ 30]: m=audio 17574 RTP/SAVP 8 0 101 [Aug 10 15:54:21] DEBUG[17897] chan_sip.c: Body 6 [ 82]: a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:i99dpczMNuFrrIJkKr09+4nfcuXTzsz5W11qjb74 [Aug 10 15:54:21] DEBUG[17897] chan_sip.c: Body 7 [ 16]: a=direction:both [Aug 10 15:54:21] DEBUG[17897] chan_sip.c: Body 8 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 10 15:54:21] DEBUG[17897] chan_sip.c: Body 9 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 10 15:54:21] DEBUG[17897] chan_sip.c: Body 10 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 10 15:54:21] DEBUG[17897] chan_sip.c: Body 11 [ 15]: a=fmtp:101 0-16 [Aug 10 15:54:21] DEBUG[17897] chan_sip.c: Body 12 [ 10]: a=ptime:20 [Aug 10 15:54:21] DEBUG[17897] chan_sip.c: Body 13 [ 10]: a=sendonly [Aug 10 15:54:21] VERBOSE[17897] chan_sip.c: --- (17 headers 14 lines) --- [Aug 10 15:54:21] DEBUG[17897] chan_sip.c: = Looking for Call ID: 3c2672a5cbc0-o7llh150d2xu (Checking From) --From tag llqi1mlu1l --To-tag as77bb01b2 [Aug 10 15:54:21] DEBUG[17897] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Aug 10 15:54:21] DEBUG[17897] netsock2.c: Splitting '192.168.2.210:2048' into... [Aug 10 15:54:21] DEBUG[17897] netsock2.c: ...host '192.168.2.210' and port '2048'. [Aug 10 15:54:21] VERBOSE[17897] chan_sip.c: Sending to 192.168.2.210:2048 (NAT) [Aug 10 15:54:21] DEBUG[17897] chan_sip.c: Initializing initreq for method INVITE - callid 3c2672a5cbc0-o7llh150d2xu [Aug 10 15:54:21] DEBUG[17897] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Aug 10 15:54:21] DEBUG[17897] chan_sip.c: Processing session-level SDP o=root 2048061303 2048061304 IN IP4 192.168.2.210... UNSUPPORTED OR FAILED. [Aug 10 15:54:21] DEBUG[17897] chan_sip.c: Processing session-level SDP s=call... UNSUPPORTED OR FAILED. [Aug 10 15:54:21] DEBUG[17897] netsock2.c: Splitting '192.168.2.210' into... [Aug 10 15:54:21] DEBUG[17897] netsock2.c: ...host '192.168.2.210' and port ''. [Aug 10 15:54:21] DEBUG[17897] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.2.210... OK. [Aug 10 15:54:21] DEBUG[17897] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Aug 10 15:54:21] VERBOSE[17897] chan_sip.c: Found RTP audio format 8 [Aug 10 15:54:21] DEBUG[17897] rtp_engine.c: Setting payload 8 based on m type on 0x416965b0 [Aug 10 15:54:21] VERBOSE[17897] chan_sip.c: Found RTP audio format 0 [Aug 10 15:54:21] DEBUG[17897] rtp_engine.c: Setting payload 0 based on m type on 0x416965b0 [Aug 10 15:54:21] VERBOSE[17897] chan_sip.c: Found RTP audio format 101 [Aug 10 15:54:21] DEBUG[17897] rtp_engine.c: Setting payload 101 based on m type on 0x416965b0 [Aug 10 15:54:21] DEBUG[17897] res_srtp.c: Adding new policy for SSRC 416744333 [Aug 10 15:54:21] DEBUG[17897] sip/sdp_crypto.c: SRTP policy activated [Aug 10 15:54:21] DEBUG[17897] chan_sip.c: Processing media-level (audio) SDP a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:i99dpczMNuFrrIJkKr09+4nfcuXTzsz5W11qjb74... OK. [Aug 10 15:54:21] DEBUG[17897] chan_sip.c: Processing media-level (audio) SDP a=direction:both... UNSUPPORTED OR FAILED. [Aug 10 15:54:21] VERBOSE[17897] chan_sip.c: Found audio description format PCMA for ID 8 [Aug 10 15:54:21] DEBUG[17897] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Aug 10 15:54:21] VERBOSE[17897] chan_sip.c: Found audio description format PCMU for ID 0 [Aug 10 15:54:21] DEBUG[17897] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Aug 10 15:54:21] VERBOSE[17897] chan_sip.c: Found audio description format telephone-event for ID 101 [Aug 10 15:54:21] DEBUG[17897] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Aug 10 15:54:21] DEBUG[17897] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED OR FAILED. [Aug 10 15:54:21] DEBUG[17897] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Aug 10 15:54:21] DEBUG[17897] chan_sip.c: Processing media-level (audio) SDP a=sendonly... OK. [Aug 10 15:54:21] DEBUG[17897] rtp_engine.c: Incorporating payload 0 on 0x416965b0 [Aug 10 15:54:21] DEBUG[17897] rtp_engine.c: Incorporating payload 8 on 0x416965b0 [Aug 10 15:54:21] DEBUG[17897] rtp_engine.c: Incorporating payload 101 on 0x416965b0 [Aug 10 15:54:21] VERBOSE[17897] chan_sip.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) [Aug 10 15:54:21] VERBOSE[17897] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Aug 10 15:54:21] DEBUG[17897] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x1c613e28' [Aug 10 15:54:21] VERBOSE[17897] chan_sip.c: Peer audio RTP is at port 192.168.2.210:17574 [Aug 10 15:54:21] DEBUG[17897] rtp_engine.c: Copying payload 0 from 0x416965b0 to 0x1c613ff0 [Aug 10 15:54:21] DEBUG[17897] rtp_engine.c: Copying payload 8 from 0x416965b0 to 0x1c613ff0 [Aug 10 15:54:21] DEBUG[17897] rtp_engine.c: Copying payload 101 from 0x416965b0 to 0x1c613ff0 [Aug 10 15:54:21] DEBUG[17897] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0x1c613e28' [Aug 10 15:54:21] DEBUG[17897] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [Aug 10 15:54:21] DEBUG[17897] chan_sip.c: We have an owner, now see if we need to change this call [Aug 10 15:54:21] DEBUG[17897] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x1c613e28' [Aug 10 15:54:21] DEBUG[17897] chan_sip.c: Got a SIP re-invite for call 3c2672a5cbc0-o7llh150d2xu [Aug 10 15:54:21] DEBUG[17897] chan_sip.c: SIP/2210-0000000b: This call is UP.... [Aug 10 15:54:21] VERBOSE[17897] chan_sip.c: <--- Transmitting (NAT) to 192.168.2.210:2048 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.2.210:2048;branch=z9hG4bK-ng8nnwgz15g4;received=192.168.2.210;rport=2048 From: "2210" ;tag=llqi1mlu1l To: ;tag=as77bb01b2 Call-ID: 3c2672a5cbc0-o7llh150d2xu CSeq: 3 INVITE Server: Asterisk PBX 1.8.15.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> [Aug 10 15:54:21] DEBUG[17897] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 192.168.2.210:2048 [Aug 10 15:54:21] DEBUG[17897] chan_sip.c: Setting framing from config on incoming call [Aug 10 15:54:21] DEBUG[17897] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: True Text flag: True [Aug 10 15:54:21] DEBUG[17897] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Aug 10 15:54:21] VERBOSE[17897] chan_sip.c: Audio is at 13020 [Aug 10 15:54:21] VERBOSE[17897] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Aug 10 15:54:21] VERBOSE[17897] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Aug 10 15:54:21] VERBOSE[17897] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 10 15:54:21] DEBUG[17897] chan_sip.c: -- Done with adding codecs to SDP [Aug 10 15:54:21] DEBUG[17897] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Aug 10 15:54:21] VERBOSE[17897] chan_sip.c: <--- Reliably Transmitting (NAT) to 192.168.2.210:2048 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.210:2048;branch=z9hG4bK-ng8nnwgz15g4;received=192.168.2.210;rport=2048 From: "2210" ;tag=llqi1mlu1l To: ;tag=as77bb01b2 Call-ID: 3c2672a5cbc0-o7llh150d2xu CSeq: 3 INVITE Server: Asterisk PBX 1.8.15.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 371 v=0 o=root 750441157 750441158 IN IP4 192.168.1.84 s=Asterisk PBX 1.8.15.0 c=IN IP4 192.168.1.84 t=0 0 m=audio 13020 RTP/SAVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=recvonly a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:9LyhOHooXaHyxyek3sEe2mCxoG81foVM00WoOcci <------------> [Aug 10 15:54:21] DEBUG[17897] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #415 [Aug 10 15:54:21] DEBUG[17897] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.2.210:2048 [Aug 10 15:54:21] DEBUG[18238] res_rtp_asterisk.c: Setting the marker bit due to a source update [Aug 10 15:54:21] VERBOSE[18238] res_musiconhold.c: -- Started music on hold, class 'default', on SIP/2212-0000000c [Aug 10 15:54:21] DEBUG[18238] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Aug 10 15:54:21] DEBUG[18238] res_rtp_asterisk.c: Setting the marker bit due to a source update [Aug 10 15:54:21] DEBUG[18238] channel.c: Got a FRAME_CONTROL (31) frame on channel SIP/2210-0000000b [Aug 10 15:54:21] DEBUG[18238] res_rtp_asterisk.c: Setting the marker bit due to a source update [Aug 10 15:54:21] DEBUG[18238] res_rtp_asterisk.c: Setting the marker bit due to a source update [Aug 10 15:54:21] DEBUG[18238] channel.c: Bridge stops bridging channels SIP/2210-0000000b and SIP/2212-0000000c [Aug 10 15:54:21] DEBUG[18238] res_rtp_asterisk.c: Setting the marker bit due to a source update [Aug 10 15:54:21] DEBUG[18238] res_rtp_asterisk.c: Setting the marker bit due to a source update [Aug 10 15:54:21] DEBUG[18238] channel.c: Generator got voice, switching to phase locked mode [Aug 10 15:54:21] DEBUG[18238] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 10 15:54:21] DEBUG[18238] channel.c: Set channel SIP/2212-0000000c to write format slin [Aug 10 15:54:21] DEBUG[18238] res_musiconhold.c: SIP/2212-0000000c Opened file 0 '/var/lib/asterisk/moh/manolo_camp-morning_coffee' [Aug 10 15:54:21] DEBUG[18238] res_rtp_asterisk.c: Difference is 1320, ms is 185 [Aug 10 15:54:21] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:21] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:21] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:21] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:21] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:21] DEBUG[17897] chan_sip.c: SIP TIMER: Rescheduling retransmission #415 (1) SIP/2.0 - 1 [Aug 10 15:54:21] DEBUG[17897] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #415)) [Aug 10 15:54:21] VERBOSE[17897] chan_sip.c: Retransmitting #1 (NAT) to 192.168.2.210:2048: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.210:2048;branch=z9hG4bK-ng8nnwgz15g4;received=192.168.2.210;rport=2048 From: "2210" ;tag=llqi1mlu1l To: ;tag=as77bb01b2 Call-ID: 3c2672a5cbc0-o7llh150d2xu CSeq: 3 INVITE Server: Asterisk PBX 1.8.15.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 371 v=0 o=root 750441157 750441158 IN IP4 192.168.1.84 s=Asterisk PBX 1.8.15.0 c=IN IP4 192.168.1.84 t=0 0 m=audio 13020 RTP/SAVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=recvonly a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:9LyhOHooXaHyxyek3sEe2mCxoG81foVM00WoOcci --- [Aug 10 15:54:21] DEBUG[17897] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.2.210:2048 [Aug 10 15:54:21] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:21] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:21] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:21] VERBOSE[17897] chan_sip.c: <--- SIP read from UDP:192.168.2.210:2048 ---> ACK sip:2212@192.168.1.84:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.210:2048;branch=z9hG4bK-plg0c1mmalks;rport From: "2210" ;tag=llqi1mlu1l To: ;tag=as77bb01b2 Call-ID: 3c2672a5cbc0-o7llh150d2xu CSeq: 3 ACK Max-Forwards: 70 Contact: ;reg-id=1 Content-Length: 0 <-------------> [Aug 10 15:54:21] DEBUG[17897] chan_sip.c: Header 0 [ 38]: ACK sip:2212@192.168.1.84:5060 SIP/2.0 [Aug 10 15:54:21] DEBUG[17897] chan_sip.c: Header 1 [ 69]: Via: SIP/2.0/UDP 192.168.2.210:2048;branch=z9hG4bK-plg0c1mmalks;rport [Aug 10 15:54:21] DEBUG[17897] chan_sip.c: Header 2 [ 51]: From: "2210" ;tag=llqi1mlu1l [Aug 10 15:54:21] DEBUG[17897] chan_sip.c: Header 3 [ 42]: To: ;tag=as77bb01b2 [Aug 10 15:54:21] DEBUG[17897] chan_sip.c: Header 4 [ 34]: Call-ID: 3c2672a5cbc0-o7llh150d2xu [Aug 10 15:54:21] DEBUG[17897] chan_sip.c: Header 5 [ 11]: CSeq: 3 ACK [Aug 10 15:54:21] DEBUG[17897] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Aug 10 15:54:21] DEBUG[17897] chan_sip.c: Header 7 [ 47]: Contact: ;reg-id=1 [Aug 10 15:54:21] DEBUG[17897] chan_sip.c: Header 8 [ 17]: Content-Length: 0 [Aug 10 15:54:21] VERBOSE[17897] chan_sip.c: --- (9 headers 0 lines) --- [Aug 10 15:54:21] DEBUG[17897] chan_sip.c: = Looking for Call ID: 3c2672a5cbc0-o7llh150d2xu (Checking From) --From tag llqi1mlu1l --To-tag as77bb01b2 [Aug 10 15:54:21] DEBUG[17897] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Aug 10 15:54:21] DEBUG[17897] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #415 [Aug 10 15:54:21] DEBUG[17897] chan_sip.c: Stopping retransmission on '3c2672a5cbc0-o7llh150d2xu' of Response 3: Match Found [Aug 10 15:54:21] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:21] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:21] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:21] VERBOSE[17897] chan_sip.c: <--- SIP read from UDP:192.168.2.210:2048 ---> ACK sip:2212@192.168.1.84:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.210:2048;branch=z9hG4bK-plg0c1mmalks;rport From: "2210" ;tag=llqi1mlu1l To: ;tag=as77bb01b2 Call-ID: 3c2672a5cbc0-o7llh150d2xu CSeq: 3 ACK Max-Forwards: 70 Contact: ;reg-id=1 Content-Length: 0 <-------------> [Aug 10 15:54:21] DEBUG[17897] chan_sip.c: Header 0 [ 38]: ACK sip:2212@192.168.1.84:5060 SIP/2.0 [Aug 10 15:54:21] DEBUG[17897] chan_sip.c: Header 1 [ 69]: Via: SIP/2.0/UDP 192.168.2.210:2048;branch=z9hG4bK-plg0c1mmalks;rport [Aug 10 15:54:21] DEBUG[17897] chan_sip.c: Header 2 [ 51]: From: "2210" ;tag=llqi1mlu1l [Aug 10 15:54:21] DEBUG[17897] chan_sip.c: Header 3 [ 42]: To: ;tag=as77bb01b2 [Aug 10 15:54:21] DEBUG[17897] chan_sip.c: Header 4 [ 34]: Call-ID: 3c2672a5cbc0-o7llh150d2xu [Aug 10 15:54:21] DEBUG[17897] chan_sip.c: Header 5 [ 11]: CSeq: 3 ACK [Aug 10 15:54:21] DEBUG[17897] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Aug 10 15:54:21] DEBUG[17897] chan_sip.c: Header 7 [ 47]: Contact: ;reg-id=1 [Aug 10 15:54:21] DEBUG[17897] chan_sip.c: Header 8 [ 17]: Content-Length: 0 [Aug 10 15:54:21] VERBOSE[17897] chan_sip.c: --- (9 headers 0 lines) --- [Aug 10 15:54:21] DEBUG[17897] chan_sip.c: = Looking for Call ID: 3c2672a5cbc0-o7llh150d2xu (Checking From) --From tag llqi1mlu1l --To-tag as77bb01b2 [Aug 10 15:54:21] DEBUG[17897] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Aug 10 15:54:21] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:21] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:21] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:21] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:21] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:21] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:21] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:21] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:21] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:21] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:21] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:22] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:22] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:22] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:22] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:22] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:22] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:22] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:22] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:22] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:22] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:22] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:22] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:22] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:22] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:22] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:22] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:22] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:22] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:22] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:22] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:22] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:22] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:22] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:22] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:22] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:22] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:22] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:22] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:22] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:22] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:22] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:22] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:22] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:22] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:22] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:22] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:22] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:22] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:22] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:22] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:22] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:22] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:22] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:22] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:22] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:22] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:22] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:22] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:22] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:22] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:23] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:23] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:23] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:23] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:23] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:23] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:23] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:23] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:23] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:23] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:23] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:23] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:23] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:23] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:23] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:23] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:23] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:23] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:23] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:23] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:23] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:23] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:23] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:23] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:23] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:23] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:23] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:23] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:23] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:23] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:23] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:23] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:23] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:23] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:23] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:23] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:23] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:23] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:23] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:23] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:23] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:23] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:23] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:23] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:23] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:23] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:23] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:23] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:23] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:23] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:24] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:24] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:24] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:24] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:24] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:24] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:24] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:24] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:24] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:24] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:24] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:24] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:24] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:24] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:24] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:24] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:24] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:24] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:24] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:24] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:24] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:24] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:24] DEBUG[17897] chan_sip.c: Allocating new SIP dialog for 0f314e770151519a44b4f291680b7eca@127.0.0.1:5060 - OPTIONS (No RTP) [Aug 10 15:54:24] DEBUG[17897] acl.c: For destination '192.168.2.210', our source address is '192.168.1.84'. [Aug 10 15:54:24] DEBUG[17897] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.1.84:5060 [Aug 10 15:54:24] DEBUG[17897] chan_sip.c: Initializing initreq for method OPTIONS - callid 028b13975c9230ca70def9f32510c563@192.168.1.84:5060 [Aug 10 15:54:24] DEBUG[17897] chan_sip.c: Header 0 [ 43]: OPTIONS sip:2210@192.168.2.210:2048 SIP/2.0 [Aug 10 15:54:24] DEBUG[17897] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK1d371c86;rport [Aug 10 15:54:24] DEBUG[17897] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 10 15:54:24] DEBUG[17897] chan_sip.c: Header 3 [ 59]: From: "asterisk" ;tag=as49f79898 [Aug 10 15:54:24] DEBUG[17897] chan_sip.c: Header 4 [ 33]: To: [Aug 10 15:54:24] DEBUG[17897] chan_sip.c: Header 5 [ 41]: Contact: [Aug 10 15:54:24] DEBUG[17897] chan_sip.c: Header 6 [ 59]: Call-ID: 028b13975c9230ca70def9f32510c563@192.168.1.84:5060 [Aug 10 15:54:24] DEBUG[17897] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:54:24] DEBUG[17897] chan_sip.c: Header 8 [ 33]: User-Agent: Asterisk PBX 1.8.15.0 [Aug 10 15:54:24] DEBUG[17897] chan_sip.c: Header 9 [ 35]: Date: Fri, 10 Aug 2012 13:54:24 GMT [Aug 10 15:54:24] DEBUG[17897] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 10 15:54:24] DEBUG[17897] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 10 15:54:24] VERBOSE[17897] chan_sip.c: Reliably Transmitting (NAT) to 192.168.2.210:2048: OPTIONS sip:2210@192.168.2.210:2048 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK1d371c86;rport Max-Forwards: 70 From: "asterisk" ;tag=as49f79898 To: Contact: Call-ID: 028b13975c9230ca70def9f32510c563@192.168.1.84:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.15.0 Date: Fri, 10 Aug 2012 13:54:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Aug 10 15:54:24] DEBUG[17897] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #416 [Aug 10 15:54:24] DEBUG[17897] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.2.210:2048 [Aug 10 15:54:24] VERBOSE[17897] chan_sip.c: <--- SIP read from UDP:192.168.2.210:2048 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK1d371c86;rport=5060 From: "asterisk" ;tag=as49f79898 To: Call-ID: 028b13975c9230ca70def9f32510c563@192.168.1.84:5060 CSeq: 102 OPTIONS Contact: ;reg-id=1 User-Agent: snom300/8.4.32 Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: 100rel, replaces, from-change Content-Length: 0 <-------------> [Aug 10 15:54:24] DEBUG[17897] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 10 15:54:24] DEBUG[17897] chan_sip.c: Header 1 [ 68]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK1d371c86;rport=5060 [Aug 10 15:54:24] DEBUG[17897] chan_sip.c: Header 2 [ 59]: From: "asterisk" ;tag=as49f79898 [Aug 10 15:54:24] DEBUG[17897] chan_sip.c: Header 3 [ 33]: To: [Aug 10 15:54:24] DEBUG[17897] chan_sip.c: Header 4 [ 59]: Call-ID: 028b13975c9230ca70def9f32510c563@192.168.1.84:5060 [Aug 10 15:54:24] DEBUG[17897] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:54:24] DEBUG[17897] chan_sip.c: Header 6 [ 47]: Contact: ;reg-id=1 [Aug 10 15:54:24] DEBUG[17897] chan_sip.c: Header 7 [ 26]: User-Agent: snom300/8.4.32 [Aug 10 15:54:24] DEBUG[17897] chan_sip.c: Header 8 [ 19]: Accept-Language: en [Aug 10 15:54:24] DEBUG[17897] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [Aug 10 15:54:24] DEBUG[17897] chan_sip.c: Header 10 [ 96]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE [Aug 10 15:54:24] DEBUG[17897] chan_sip.c: Header 11 [ 42]: Allow-Events: talk, hold, refer, call-info [Aug 10 15:54:24] DEBUG[17897] chan_sip.c: Header 12 [ 40]: Supported: 100rel, replaces, from-change [Aug 10 15:54:24] DEBUG[17897] chan_sip.c: Header 13 [ 17]: Content-Length: 0 [Aug 10 15:54:24] VERBOSE[17897] chan_sip.c: --- (14 headers 0 lines) --- [Aug 10 15:54:24] DEBUG[17897] chan_sip.c: = Looking for Call ID: 028b13975c9230ca70def9f32510c563@192.168.1.84:5060 (Checking To) --From tag as49f79898 --To-tag [Aug 10 15:54:24] DEBUG[17897] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #416 [Aug 10 15:54:24] DEBUG[17897] chan_sip.c: Stopping retransmission on '028b13975c9230ca70def9f32510c563@192.168.1.84:5060' of Request 102: Match Found [Aug 10 15:54:24] DEBUG[17897] chan_sip.c: Destroying SIP dialog 028b13975c9230ca70def9f32510c563@192.168.1.84:5060 [Aug 10 15:54:24] VERBOSE[17897] chan_sip.c: Really destroying SIP dialog '028b13975c9230ca70def9f32510c563@192.168.1.84:5060' Method: OPTIONS [Aug 10 15:54:24] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:24] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:24] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:24] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:24] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:24] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:24] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:24] DEBUG[17897] chan_sip.c: Allocating new SIP dialog for 4b4603e43008685a7248f86d74364ca9@127.0.0.1:5060 - OPTIONS (No RTP) [Aug 10 15:54:24] DEBUG[17897] acl.c: For destination '192.168.1.102', our source address is '192.168.1.84'. [Aug 10 15:54:24] DEBUG[17897] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.1.84:5060 [Aug 10 15:54:24] DEBUG[17897] chan_sip.c: Initializing initreq for method OPTIONS - callid 1db133407b2a4622513bb5f4020a00f3@192.168.1.84:5060 [Aug 10 15:54:24] DEBUG[17897] chan_sip.c: Header 0 [ 57]: OPTIONS sip:2212@192.168.1.102:2048;line=z1zv7jsj SIP/2.0 [Aug 10 15:54:24] DEBUG[17897] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK403718b7;rport [Aug 10 15:54:24] DEBUG[17897] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 10 15:54:24] DEBUG[17897] chan_sip.c: Header 3 [ 59]: From: "asterisk" ;tag=as00e59d19 [Aug 10 15:54:24] DEBUG[17897] chan_sip.c: Header 4 [ 47]: To: [Aug 10 15:54:24] DEBUG[17897] chan_sip.c: Header 5 [ 41]: Contact: [Aug 10 15:54:24] DEBUG[17897] chan_sip.c: Header 6 [ 59]: Call-ID: 1db133407b2a4622513bb5f4020a00f3@192.168.1.84:5060 [Aug 10 15:54:24] DEBUG[17897] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:54:24] DEBUG[17897] chan_sip.c: Header 8 [ 33]: User-Agent: Asterisk PBX 1.8.15.0 [Aug 10 15:54:24] DEBUG[17897] chan_sip.c: Header 9 [ 35]: Date: Fri, 10 Aug 2012 13:54:24 GMT [Aug 10 15:54:24] DEBUG[17897] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 10 15:54:24] DEBUG[17897] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 10 15:54:24] VERBOSE[17897] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.102:2048: OPTIONS sip:2212@192.168.1.102:2048;line=z1zv7jsj SIP/2.0 Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK403718b7;rport Max-Forwards: 70 From: "asterisk" ;tag=as00e59d19 To: Contact: Call-ID: 1db133407b2a4622513bb5f4020a00f3@192.168.1.84:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.15.0 Date: Fri, 10 Aug 2012 13:54:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Aug 10 15:54:24] DEBUG[17897] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #419 [Aug 10 15:54:24] DEBUG[17897] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.1.102:2048 [Aug 10 15:54:24] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:24] VERBOSE[17897] chan_sip.c: <--- SIP read from UDP:192.168.1.102:2048 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK403718b7;rport=5060 From: "asterisk" ;tag=as00e59d19 To: Call-ID: 1db133407b2a4622513bb5f4020a00f3@192.168.1.84:5060 CSeq: 102 OPTIONS Contact: ;reg-id=1 User-Agent: snom300/8.4.32 Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Content-Length: 0 <-------------> [Aug 10 15:54:24] DEBUG[17897] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 10 15:54:24] DEBUG[17897] chan_sip.c: Header 1 [ 68]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK403718b7;rport=5060 [Aug 10 15:54:24] DEBUG[17897] chan_sip.c: Header 2 [ 59]: From: "asterisk" ;tag=as00e59d19 [Aug 10 15:54:24] DEBUG[17897] chan_sip.c: Header 3 [ 47]: To: [Aug 10 15:54:24] DEBUG[17897] chan_sip.c: Header 4 [ 59]: Call-ID: 1db133407b2a4622513bb5f4020a00f3@192.168.1.84:5060 [Aug 10 15:54:24] DEBUG[17897] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:54:24] DEBUG[17897] chan_sip.c: Header 6 [ 61]: Contact: ;reg-id=1 [Aug 10 15:54:24] DEBUG[17897] chan_sip.c: Header 7 [ 26]: User-Agent: snom300/8.4.32 [Aug 10 15:54:24] DEBUG[17897] chan_sip.c: Header 8 [ 19]: Accept-Language: en [Aug 10 15:54:24] DEBUG[17897] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [Aug 10 15:54:24] DEBUG[17897] chan_sip.c: Header 10 [ 96]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE [Aug 10 15:54:24] DEBUG[17897] chan_sip.c: Header 11 [ 42]: Allow-Events: talk, hold, refer, call-info [Aug 10 15:54:24] DEBUG[17897] chan_sip.c: Header 12 [ 47]: Supported: timer, 100rel, replaces, from-change [Aug 10 15:54:24] DEBUG[17897] chan_sip.c: Header 13 [ 17]: Content-Length: 0 [Aug 10 15:54:24] VERBOSE[17897] chan_sip.c: --- (14 headers 0 lines) --- [Aug 10 15:54:24] DEBUG[17897] chan_sip.c: = Looking for Call ID: 1db133407b2a4622513bb5f4020a00f3@192.168.1.84:5060 (Checking To) --From tag as00e59d19 --To-tag [Aug 10 15:54:24] DEBUG[17897] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #419 [Aug 10 15:54:24] DEBUG[17897] chan_sip.c: Stopping retransmission on '1db133407b2a4622513bb5f4020a00f3@192.168.1.84:5060' of Request 102: Match Found [Aug 10 15:54:24] DEBUG[17897] chan_sip.c: Destroying SIP dialog 1db133407b2a4622513bb5f4020a00f3@192.168.1.84:5060 [Aug 10 15:54:24] VERBOSE[17897] chan_sip.c: Really destroying SIP dialog '1db133407b2a4622513bb5f4020a00f3@192.168.1.84:5060' Method: OPTIONS [Aug 10 15:54:24] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:24] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:24] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:24] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:24] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:24] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:24] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:24] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:24] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:24] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:24] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:54:24] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:24] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:24] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:24] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:24] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:24] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:24] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:24] DEBUG[18238] res_rtp_asterisk.c: RTCP NAT: Got RTCP from other end. Now sending to address 192.168.2.210:17575 [Aug 10 15:54:24] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 24 bytes [Aug 10 15:54:24] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:24] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:24] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:25] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:25] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:25] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:25] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:25] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:25] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:25] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:25] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:25] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:25] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:25] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:25] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:25] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:25] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:25] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:25] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:25] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:25] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:25] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:25] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:25] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:25] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:25] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:25] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:25] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:25] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:25] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:25] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:25] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:25] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:25] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:25] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:25] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:25] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:25] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:25] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:25] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:25] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:25] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:25] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:25] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:25] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:25] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:25] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:25] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:25] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:25] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:25] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:25] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:25] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:26] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:26] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:26] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:26] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:26] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:26] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:26] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:26] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:26] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:26] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:26] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:26] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:26] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:26] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:26] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:26] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:26] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:26] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:26] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:26] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:26] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:26] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:26] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:26] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:26] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:26] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:26] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:26] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:26] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:26] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:26] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:26] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:26] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:26] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:26] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:26] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:26] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:26] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:26] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:26] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:26] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:26] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:26] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:26] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:26] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:26] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:26] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:26] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:26] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:26] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:27] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:27] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:27] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:27] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:27] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:27] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:27] DEBUG[18238] res_rtp_asterisk.c: No remote address on RTP instance '0x1c613e28' so dropping frame [Aug 10 15:54:27] VERBOSE[17897] chan_sip.c: <--- SIP read from UDP:192.168.2.210:2048 ---> INVITE sip:2212@192.168.1.84:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.210:2048;branch=z9hG4bK-2bmcwmwtnjfi;rport From: "2210" ;tag=llqi1mlu1l To: ;tag=as77bb01b2 Call-ID: 3c2672a5cbc0-o7llh150d2xu CSeq: 4 INVITE Max-Forwards: 70 Contact: ;reg-id=1 X-Serialnumber: 0004132500A7 P-Key-Flags: keys="3" User-Agent: snom300/8.4.32 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: 100rel, replaces, from-change Content-Type: application/sdp Content-Length: 349 v=0 o=root 2048061303 2048061305 IN IP4 192.168.2.210 s=call c=IN IP4 192.168.2.210 t=0 0 m=audio 17574 RTP/SAVP 8 0 101 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:i99dpczMNuFrrIJkKr09+4nfcuXTzsz5W11qjb74 a=direction:both a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <-------------> [Aug 10 15:54:27] DEBUG[17897] chan_sip.c: Header 0 [ 41]: INVITE sip:2212@192.168.1.84:5060 SIP/2.0 [Aug 10 15:54:27] DEBUG[17897] chan_sip.c: Header 1 [ 69]: Via: SIP/2.0/UDP 192.168.2.210:2048;branch=z9hG4bK-2bmcwmwtnjfi;rport [Aug 10 15:54:27] DEBUG[17897] chan_sip.c: Header 2 [ 51]: From: "2210" ;tag=llqi1mlu1l [Aug 10 15:54:27] DEBUG[17897] chan_sip.c: Header 3 [ 42]: To: ;tag=as77bb01b2 [Aug 10 15:54:27] DEBUG[17897] chan_sip.c: Header 4 [ 34]: Call-ID: 3c2672a5cbc0-o7llh150d2xu [Aug 10 15:54:27] DEBUG[17897] chan_sip.c: Header 5 [ 14]: CSeq: 4 INVITE [Aug 10 15:54:27] DEBUG[17897] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Aug 10 15:54:27] DEBUG[17897] chan_sip.c: Header 7 [ 47]: Contact: ;reg-id=1 [Aug 10 15:54:27] DEBUG[17897] chan_sip.c: Header 8 [ 28]: X-Serialnumber: 0004132500A7 [Aug 10 15:54:27] DEBUG[17897] chan_sip.c: Header 9 [ 21]: P-Key-Flags: keys="3" [Aug 10 15:54:27] DEBUG[17897] chan_sip.c: Header 10 [ 26]: User-Agent: snom300/8.4.32 [Aug 10 15:54:27] DEBUG[17897] chan_sip.c: Header 11 [ 23]: Accept: application/sdp [Aug 10 15:54:27] DEBUG[17897] chan_sip.c: Header 12 [ 96]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE [Aug 10 15:54:27] DEBUG[17897] chan_sip.c: Header 13 [ 42]: Allow-Events: talk, hold, refer, call-info [Aug 10 15:54:27] DEBUG[17897] chan_sip.c: Header 14 [ 40]: Supported: 100rel, replaces, from-change [Aug 10 15:54:27] DEBUG[17897] chan_sip.c: Header 15 [ 29]: Content-Type: application/sdp [Aug 10 15:54:27] DEBUG[17897] chan_sip.c: Header 16 [ 19]: Content-Length: 349 [Aug 10 15:54:27] DEBUG[17897] chan_sip.c: Header 17 [ 0]: [Aug 10 15:54:27] DEBUG[17897] chan_sip.c: Body 0 [ 3]: v=0 [Aug 10 15:54:27] DEBUG[17897] chan_sip.c: Body 1 [ 49]: o=root 2048061303 2048061305 IN IP4 192.168.2.210 [Aug 10 15:54:27] DEBUG[17897] chan_sip.c: Body 2 [ 6]: s=call [Aug 10 15:54:27] DEBUG[17897] chan_sip.c: Body 3 [ 22]: c=IN IP4 192.168.2.210 [Aug 10 15:54:27] DEBUG[17897] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 10 15:54:27] DEBUG[17897] chan_sip.c: Body 5 [ 30]: m=audio 17574 RTP/SAVP 8 0 101 [Aug 10 15:54:27] DEBUG[17897] chan_sip.c: Body 6 [ 82]: a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:i99dpczMNuFrrIJkKr09+4nfcuXTzsz5W11qjb74 [Aug 10 15:54:27] DEBUG[17897] chan_sip.c: Body 7 [ 16]: a=direction:both [Aug 10 15:54:27] DEBUG[17897] chan_sip.c: Body 8 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 10 15:54:27] DEBUG[17897] chan_sip.c: Body 9 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 10 15:54:27] DEBUG[17897] chan_sip.c: Body 10 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 10 15:54:27] DEBUG[17897] chan_sip.c: Body 11 [ 15]: a=fmtp:101 0-16 [Aug 10 15:54:27] DEBUG[17897] chan_sip.c: Body 12 [ 10]: a=ptime:20 [Aug 10 15:54:27] DEBUG[17897] chan_sip.c: Body 13 [ 10]: a=sendrecv [Aug 10 15:54:27] VERBOSE[17897] chan_sip.c: --- (17 headers 14 lines) --- [Aug 10 15:54:27] DEBUG[17897] chan_sip.c: = Looking for Call ID: 3c2672a5cbc0-o7llh150d2xu (Checking From) --From tag llqi1mlu1l --To-tag as77bb01b2 [Aug 10 15:54:27] DEBUG[17897] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Aug 10 15:54:27] DEBUG[17897] netsock2.c: Splitting '192.168.2.210:2048' into... [Aug 10 15:54:27] DEBUG[17897] netsock2.c: ...host '192.168.2.210' and port '2048'. [Aug 10 15:54:27] VERBOSE[17897] chan_sip.c: Sending to 192.168.2.210:2048 (NAT) [Aug 10 15:54:27] DEBUG[17897] chan_sip.c: Initializing initreq for method INVITE - callid 3c2672a5cbc0-o7llh150d2xu [Aug 10 15:54:27] DEBUG[17897] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Aug 10 15:54:27] DEBUG[17897] chan_sip.c: Processing session-level SDP o=root 2048061303 2048061305 IN IP4 192.168.2.210... UNSUPPORTED OR FAILED. [Aug 10 15:54:27] DEBUG[17897] chan_sip.c: Processing session-level SDP s=call... UNSUPPORTED OR FAILED. [Aug 10 15:54:27] DEBUG[17897] netsock2.c: Splitting '192.168.2.210' into... [Aug 10 15:54:27] DEBUG[17897] netsock2.c: ...host '192.168.2.210' and port ''. [Aug 10 15:54:27] DEBUG[17897] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.2.210... OK. [Aug 10 15:54:27] DEBUG[17897] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Aug 10 15:54:27] VERBOSE[17897] chan_sip.c: Found RTP audio format 8 [Aug 10 15:54:27] DEBUG[17897] rtp_engine.c: Setting payload 8 based on m type on 0x416965b0 [Aug 10 15:54:27] VERBOSE[17897] chan_sip.c: Found RTP audio format 0 [Aug 10 15:54:27] DEBUG[17897] rtp_engine.c: Setting payload 0 based on m type on 0x416965b0 [Aug 10 15:54:27] VERBOSE[17897] chan_sip.c: Found RTP audio format 101 [Aug 10 15:54:27] DEBUG[17897] rtp_engine.c: Setting payload 101 based on m type on 0x416965b0 [Aug 10 15:54:27] DEBUG[17897] res_srtp.c: Adding new policy for SSRC 416744333 [Aug 10 15:54:27] DEBUG[17897] sip/sdp_crypto.c: SRTP policy activated [Aug 10 15:54:27] DEBUG[17897] chan_sip.c: Processing media-level (audio) SDP a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:i99dpczMNuFrrIJkKr09+4nfcuXTzsz5W11qjb74... OK. [Aug 10 15:54:27] DEBUG[17897] chan_sip.c: Processing media-level (audio) SDP a=direction:both... UNSUPPORTED OR FAILED. [Aug 10 15:54:27] VERBOSE[17897] chan_sip.c: Found audio description format PCMA for ID 8 [Aug 10 15:54:27] DEBUG[17897] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Aug 10 15:54:27] VERBOSE[17897] chan_sip.c: Found audio description format PCMU for ID 0 [Aug 10 15:54:27] DEBUG[17897] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Aug 10 15:54:27] VERBOSE[17897] chan_sip.c: Found audio description format telephone-event for ID 101 [Aug 10 15:54:27] DEBUG[17897] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Aug 10 15:54:27] DEBUG[17897] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED OR FAILED. [Aug 10 15:54:27] DEBUG[17897] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Aug 10 15:54:27] DEBUG[17897] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Aug 10 15:54:27] DEBUG[17897] rtp_engine.c: Incorporating payload 0 on 0x416965b0 [Aug 10 15:54:27] DEBUG[17897] rtp_engine.c: Incorporating payload 8 on 0x416965b0 [Aug 10 15:54:27] DEBUG[17897] rtp_engine.c: Incorporating payload 101 on 0x416965b0 [Aug 10 15:54:27] VERBOSE[17897] chan_sip.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) [Aug 10 15:54:27] VERBOSE[17897] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Aug 10 15:54:27] DEBUG[17897] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x1c613e28' [Aug 10 15:54:27] VERBOSE[17897] chan_sip.c: Peer audio RTP is at port 192.168.2.210:17574 [Aug 10 15:54:27] DEBUG[17897] rtp_engine.c: Copying payload 0 from 0x416965b0 to 0x1c613ff0 [Aug 10 15:54:27] DEBUG[17897] rtp_engine.c: Copying payload 8 from 0x416965b0 to 0x1c613ff0 [Aug 10 15:54:27] DEBUG[17897] rtp_engine.c: Copying payload 101 from 0x416965b0 to 0x1c613ff0 [Aug 10 15:54:27] DEBUG[17897] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0x1c613e28' [Aug 10 15:54:27] DEBUG[17897] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [Aug 10 15:54:27] DEBUG[17897] chan_sip.c: We have an owner, now see if we need to change this call [Aug 10 15:54:27] DEBUG[17897] chan_sip.c: Got a SIP re-invite for call 3c2672a5cbc0-o7llh150d2xu [Aug 10 15:54:27] DEBUG[17897] chan_sip.c: SIP/2210-0000000b: This call is UP.... [Aug 10 15:54:27] VERBOSE[17897] chan_sip.c: <--- Transmitting (NAT) to 192.168.2.210:2048 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.2.210:2048;branch=z9hG4bK-2bmcwmwtnjfi;received=192.168.2.210;rport=2048 From: "2210" ;tag=llqi1mlu1l To: ;tag=as77bb01b2 Call-ID: 3c2672a5cbc0-o7llh150d2xu CSeq: 4 INVITE Server: Asterisk PBX 1.8.15.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> [Aug 10 15:54:27] DEBUG[17897] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 192.168.2.210:2048 [Aug 10 15:54:27] DEBUG[17897] chan_sip.c: Setting framing from config on incoming call [Aug 10 15:54:27] DEBUG[17897] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: True Text flag: True [Aug 10 15:54:27] DEBUG[17897] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Aug 10 15:54:27] VERBOSE[17897] chan_sip.c: Audio is at 13020 [Aug 10 15:54:27] VERBOSE[17897] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Aug 10 15:54:27] VERBOSE[17897] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Aug 10 15:54:27] VERBOSE[17897] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 10 15:54:27] DEBUG[17897] chan_sip.c: -- Done with adding codecs to SDP [Aug 10 15:54:27] DEBUG[17897] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Aug 10 15:54:27] VERBOSE[17897] chan_sip.c: <--- Reliably Transmitting (NAT) to 192.168.2.210:2048 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.210:2048;branch=z9hG4bK-2bmcwmwtnjfi;received=192.168.2.210;rport=2048 From: "2210" ;tag=llqi1mlu1l To: ;tag=as77bb01b2 Call-ID: 3c2672a5cbc0-o7llh150d2xu CSeq: 4 INVITE Server: Asterisk PBX 1.8.15.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 371 v=0 o=root 750441157 750441159 IN IP4 192.168.1.84 s=Asterisk PBX 1.8.15.0 c=IN IP4 192.168.1.84 t=0 0 m=audio 13020 RTP/SAVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:9LyhOHooXaHyxyek3sEe2mCxoG81foVM00WoOcci <------------> [Aug 10 15:54:27] DEBUG[17897] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #422 [Aug 10 15:54:27] DEBUG[17897] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.2.210:2048 [Aug 10 15:54:27] DEBUG[18238] res_rtp_asterisk.c: Setting the marker bit due to a source update [Aug 10 15:54:27] VERBOSE[18238] res_musiconhold.c: -- Stopped music on hold on SIP/2212-0000000c [Aug 10 15:54:27] DEBUG[18238] channel.c: Set channel SIP/2212-0000000c to write format alaw [Aug 10 15:54:27] DEBUG[18238] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 10 15:54:27] DEBUG[18238] res_rtp_asterisk.c: Setting the marker bit due to a source update [Aug 10 15:54:27] DEBUG[18238] channel.c: Got a FRAME_CONTROL (31) frame on channel SIP/2210-0000000b [Aug 10 15:54:27] DEBUG[18238] res_rtp_asterisk.c: Setting the marker bit due to a source update [Aug 10 15:54:27] DEBUG[18238] res_rtp_asterisk.c: Setting the marker bit due to a source update [Aug 10 15:54:27] DEBUG[18238] channel.c: Bridge stops bridging channels SIP/2210-0000000b and SIP/2212-0000000c [Aug 10 15:54:27] DEBUG[18238] res_rtp_asterisk.c: Setting the marker bit due to a source update [Aug 10 15:54:27] DEBUG[18238] res_rtp_asterisk.c: Setting the marker bit due to a source update [Aug 10 15:54:27] DEBUG[18238] res_rtp_asterisk.c: Starting RTCP transmission on RTP instance '0x1c613e28' [Aug 10 15:54:27] DEBUG[17897] chan_sip.c: SIP TIMER: Rescheduling retransmission #422 (1) SIP/2.0 - 1 [Aug 10 15:54:27] DEBUG[17897] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #422)) [Aug 10 15:54:27] VERBOSE[17897] chan_sip.c: Retransmitting #1 (NAT) to 192.168.2.210:2048: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.210:2048;branch=z9hG4bK-2bmcwmwtnjfi;received=192.168.2.210;rport=2048 From: "2210" ;tag=llqi1mlu1l To: ;tag=as77bb01b2 Call-ID: 3c2672a5cbc0-o7llh150d2xu CSeq: 4 INVITE Server: Asterisk PBX 1.8.15.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 371 v=0 o=root 750441157 750441159 IN IP4 192.168.1.84 s=Asterisk PBX 1.8.15.0 c=IN IP4 192.168.1.84 t=0 0 m=audio 13020 RTP/SAVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:9LyhOHooXaHyxyek3sEe2mCxoG81foVM00WoOcci --- [Aug 10 15:54:27] DEBUG[17897] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.2.210:2048 [Aug 10 15:54:27] VERBOSE[17897] chan_sip.c: <--- SIP read from UDP:192.168.2.210:2048 ---> ACK sip:2212@192.168.1.84:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.210:2048;branch=z9hG4bK-4zukc7dzu59i;rport From: "2210" ;tag=llqi1mlu1l To: ;tag=as77bb01b2 Call-ID: 3c2672a5cbc0-o7llh150d2xu CSeq: 4 ACK Max-Forwards: 70 Contact: ;reg-id=1 Content-Length: 0 <-------------> [Aug 10 15:54:27] DEBUG[17897] chan_sip.c: Header 0 [ 38]: ACK sip:2212@192.168.1.84:5060 SIP/2.0 [Aug 10 15:54:27] DEBUG[17897] chan_sip.c: Header 1 [ 69]: Via: SIP/2.0/UDP 192.168.2.210:2048;branch=z9hG4bK-4zukc7dzu59i;rport [Aug 10 15:54:27] DEBUG[17897] chan_sip.c: Header 2 [ 51]: From: "2210" ;tag=llqi1mlu1l [Aug 10 15:54:27] DEBUG[17897] chan_sip.c: Header 3 [ 42]: To: ;tag=as77bb01b2 [Aug 10 15:54:27] DEBUG[17897] chan_sip.c: Header 4 [ 34]: Call-ID: 3c2672a5cbc0-o7llh150d2xu [Aug 10 15:54:27] DEBUG[17897] chan_sip.c: Header 5 [ 11]: CSeq: 4 ACK [Aug 10 15:54:27] DEBUG[17897] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Aug 10 15:54:27] DEBUG[17897] chan_sip.c: Header 7 [ 47]: Contact: ;reg-id=1 [Aug 10 15:54:27] DEBUG[17897] chan_sip.c: Header 8 [ 17]: Content-Length: 0 [Aug 10 15:54:27] VERBOSE[17897] chan_sip.c: --- (9 headers 0 lines) --- [Aug 10 15:54:27] DEBUG[17897] chan_sip.c: = Looking for Call ID: 3c2672a5cbc0-o7llh150d2xu (Checking From) --From tag llqi1mlu1l --To-tag as77bb01b2 [Aug 10 15:54:27] DEBUG[17897] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Aug 10 15:54:27] DEBUG[17897] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #422 [Aug 10 15:54:27] DEBUG[17897] chan_sip.c: Stopping retransmission on '3c2672a5cbc0-o7llh150d2xu' of Response 4: Match Found [Aug 10 15:54:27] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:54:27] VERBOSE[17897] chan_sip.c: <--- SIP read from UDP:192.168.2.210:2048 ---> ACK sip:2212@192.168.1.84:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.210:2048;branch=z9hG4bK-4zukc7dzu59i;rport From: "2210" ;tag=llqi1mlu1l To: ;tag=as77bb01b2 Call-ID: 3c2672a5cbc0-o7llh150d2xu CSeq: 4 ACK Max-Forwards: 70 Contact: ;reg-id=1 Content-Length: 0 <-------------> [Aug 10 15:54:27] DEBUG[17897] chan_sip.c: Header 0 [ 38]: ACK sip:2212@192.168.1.84:5060 SIP/2.0 [Aug 10 15:54:27] DEBUG[17897] chan_sip.c: Header 1 [ 69]: Via: SIP/2.0/UDP 192.168.2.210:2048;branch=z9hG4bK-4zukc7dzu59i;rport [Aug 10 15:54:27] DEBUG[17897] chan_sip.c: Header 2 [ 51]: From: "2210" ;tag=llqi1mlu1l [Aug 10 15:54:27] DEBUG[17897] chan_sip.c: Header 3 [ 42]: To: ;tag=as77bb01b2 [Aug 10 15:54:27] DEBUG[17897] chan_sip.c: Header 4 [ 34]: Call-ID: 3c2672a5cbc0-o7llh150d2xu [Aug 10 15:54:27] DEBUG[17897] chan_sip.c: Header 5 [ 11]: CSeq: 4 ACK [Aug 10 15:54:27] DEBUG[17897] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Aug 10 15:54:27] DEBUG[17897] chan_sip.c: Header 7 [ 47]: Contact: ;reg-id=1 [Aug 10 15:54:27] DEBUG[17897] chan_sip.c: Header 8 [ 17]: Content-Length: 0 [Aug 10 15:54:27] VERBOSE[17897] chan_sip.c: --- (9 headers 0 lines) --- [Aug 10 15:54:27] DEBUG[17897] chan_sip.c: = Looking for Call ID: 3c2672a5cbc0-o7llh150d2xu (Checking From) --From tag llqi1mlu1l --To-tag as77bb01b2 [Aug 10 15:54:27] DEBUG[17897] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Aug 10 15:54:27] WARNING[18238] res_srtp.c: SRTP unprotect failed with: authentication failure 10 [Aug 10 15:54:29] WARNING[18238] res_srtp.c: SRTP unprotect failed with: authentication failure 110 [Aug 10 15:54:29] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:54:31] WARNING[18238] res_srtp.c: SRTP unprotect failed with: authentication failure 110 [Aug 10 15:54:32] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:54:33] WARNING[18238] res_srtp.c: SRTP unprotect failed with: authentication failure 110 [Aug 10 15:54:34] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:54:35] WARNING[18238] res_srtp.c: SRTP unprotect failed with: authentication failure 110 [Aug 10 15:54:37] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:54:37] WARNING[18238] res_srtp.c: SRTP unprotect failed with: authentication failure 110 [Aug 10 15:54:39] WARNING[18238] res_srtp.c: SRTP unprotect failed with: authentication failure 110 [Aug 10 15:54:39] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:54:41] WARNING[18238] res_srtp.c: SRTP unprotect failed with: authentication failure 110 [Aug 10 15:54:42] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:54:43] WARNING[18238] res_srtp.c: SRTP unprotect failed with: authentication failure 110 [Aug 10 15:54:44] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:54:45] WARNING[18238] res_srtp.c: SRTP unprotect failed with: authentication failure 110 [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: Allocating new SIP dialog for 2b92e8810fcbe9de50e7f9ea24c2ae83@127.0.0.1:5060 - OPTIONS (No RTP) [Aug 10 15:54:46] DEBUG[17897] acl.c: For destination '192.168.1.106', our source address is '192.168.1.84'. [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.1.84:5060 [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: Initializing initreq for method OPTIONS - callid 7c676d5e1e0c4d8122f1287b4a96affc@192.168.1.84:5060 [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: Header 0 [ 57]: OPTIONS sip:2219@192.168.1.106:2048;line=7g8ggsvj SIP/2.0 [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK46c3336c;rport [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: Header 3 [ 59]: From: "asterisk" ;tag=as6ae43ceb [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: Header 4 [ 47]: To: [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: Header 5 [ 41]: Contact: [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: Header 6 [ 59]: Call-ID: 7c676d5e1e0c4d8122f1287b4a96affc@192.168.1.84:5060 [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: Header 8 [ 33]: User-Agent: Asterisk PBX 1.8.15.0 [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: Header 9 [ 35]: Date: Fri, 10 Aug 2012 13:54:46 GMT [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 10 15:54:46] VERBOSE[17897] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.106:2048: OPTIONS sip:2219@192.168.1.106:2048;line=7g8ggsvj SIP/2.0 Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK46c3336c;rport Max-Forwards: 70 From: "asterisk" ;tag=as6ae43ceb To: Contact: Call-ID: 7c676d5e1e0c4d8122f1287b4a96affc@192.168.1.84:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.15.0 Date: Fri, 10 Aug 2012 13:54:46 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #424 [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.1.106:2048 [Aug 10 15:54:46] VERBOSE[17897] chan_sip.c: <--- SIP read from UDP:192.168.1.106:2048 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK46c3336c;rport=5060 From: "asterisk" ;tag=as6ae43ceb To: ;tag=uihxzeegfb Call-ID: 7c676d5e1e0c4d8122f1287b4a96affc@192.168.1.84:5060 CSeq: 102 OPTIONS Contact: ;reg-id=1 User-Agent: snom300/8.7.3.10 Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: 100rel, replaces, from-change Content-Length: 0 <-------------> [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: Header 1 [ 68]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK46c3336c;rport=5060 [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: Header 2 [ 59]: From: "asterisk" ;tag=as6ae43ceb [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: Header 3 [ 62]: To: ;tag=uihxzeegfb [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: Header 4 [ 59]: Call-ID: 7c676d5e1e0c4d8122f1287b4a96affc@192.168.1.84:5060 [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: Header 6 [ 61]: Contact: ;reg-id=1 [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: Header 7 [ 28]: User-Agent: snom300/8.7.3.10 [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: Header 8 [ 19]: Accept-Language: en [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: Header 10 [ 96]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: Header 11 [ 42]: Allow-Events: talk, hold, refer, call-info [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: Header 12 [ 40]: Supported: 100rel, replaces, from-change [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: Header 13 [ 17]: Content-Length: 0 [Aug 10 15:54:46] VERBOSE[17897] chan_sip.c: --- (14 headers 0 lines) --- [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: = Looking for Call ID: 7c676d5e1e0c4d8122f1287b4a96affc@192.168.1.84:5060 (Checking To) --From tag as6ae43ceb --To-tag uihxzeegfb [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #424 [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: Stopping retransmission on '7c676d5e1e0c4d8122f1287b4a96affc@192.168.1.84:5060' of Request 102: Match Found [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: Destroying SIP dialog 7c676d5e1e0c4d8122f1287b4a96affc@192.168.1.84:5060 [Aug 10 15:54:46] VERBOSE[17897] chan_sip.c: Really destroying SIP dialog '7c676d5e1e0c4d8122f1287b4a96affc@192.168.1.84:5060' Method: OPTIONS [Aug 10 15:54:46] VERBOSE[17897] chan_sip.c: <--- SIP read from UDP:192.168.1.106:2048 ---> REGISTER sip:192.168.1.84 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.106:2048;branch=z9hG4bK-5n1gw2w9djw7;rport From: "2219" ;tag=r7jeyhhxyv To: "2219" Call-ID: 50250e9b9684-8d3g9c7o4hy2 CSeq: 41 REGISTER Max-Forwards: 70 Contact: ;reg-id=1;q=1.0;+sip.instance="";audio;mobility="fixed";duplex="full";description="snom300";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO" User-Agent: snom300/8.7.3.10 Allow-Events: dialog X-Real-IP: 192.168.1.106 Supported: path, gruu Content-Length: 0 <-------------> [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: Header 0 [ 33]: REGISTER sip:192.168.1.84 SIP/2.0 [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: Header 1 [ 69]: Via: SIP/2.0/UDP 192.168.1.106:2048;branch=z9hG4bK-5n1gw2w9djw7;rport [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: Header 2 [ 51]: From: "2219" ;tag=r7jeyhhxyv [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: Header 3 [ 34]: To: "2219" [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: Header 4 [ 34]: Call-ID: 50250e9b9684-8d3g9c7o4hy2 [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: Header 5 [ 17]: CSeq: 41 REGISTER [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: Header 7 [306]: Contact: ;reg-id=1;q=1.0;+sip.instance="";audio;mobility="fixed";duplex="full";description="snom300";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO" [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: Header 8 [ 28]: User-Agent: snom300/8.7.3.10 [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: Header 9 [ 20]: Allow-Events: dialog [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: Header 10 [ 24]: X-Real-IP: 192.168.1.106 [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: Header 11 [ 21]: Supported: path, gruu [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Aug 10 15:54:46] VERBOSE[17897] chan_sip.c: --- (13 headers 0 lines) --- [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: = Looking for Call ID: 50250e9b9684-8d3g9c7o4hy2 (Checking From) --From tag r7jeyhhxyv --To-tag [Aug 10 15:54:46] DEBUG[17897] acl.c: For destination '192.168.1.106', our source address is '192.168.1.84'. [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.1.84:5060 [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: Allocating new SIP dialog for 50250e9b9684-8d3g9c7o4hy2 - REGISTER (No RTP) [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: Initializing initreq for method REGISTER - callid 50250e9b9684-8d3g9c7o4hy2 [Aug 10 15:54:46] DEBUG[17897] netsock2.c: Splitting '192.168.1.106:2048' into... [Aug 10 15:54:46] DEBUG[17897] netsock2.c: ...host '192.168.1.106' and port '2048'. [Aug 10 15:54:46] VERBOSE[17897] chan_sip.c: Sending to 192.168.1.106:2048 (NAT) [Aug 10 15:54:46] DEBUG[17897] netsock2.c: Splitting '192.168.1.84' into... [Aug 10 15:54:46] DEBUG[17897] netsock2.c: ...host '192.168.1.84' and port ''. [Aug 10 15:54:46] VERBOSE[17897] chan_sip.c: <--- Transmitting (NAT) to 192.168.1.106:2048 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.106:2048;branch=z9hG4bK-5n1gw2w9djw7;received=192.168.1.106;rport=2048 From: "2219" ;tag=r7jeyhhxyv To: "2219" ;tag=as0317aa01 Call-ID: 50250e9b9684-8d3g9c7o4hy2 CSeq: 41 REGISTER Server: Asterisk PBX 1.8.15.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5d67a532" Content-Length: 0 <------------> [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 192.168.1.106:2048 [Aug 10 15:54:46] VERBOSE[17897] chan_sip.c: Scheduling destruction of SIP dialog '50250e9b9684-8d3g9c7o4hy2' in 32000 ms (Method: REGISTER) [Aug 10 15:54:46] VERBOSE[17897] chan_sip.c: <--- SIP read from UDP:192.168.1.106:2048 ---> REGISTER sip:192.168.1.84 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.106:2048;branch=z9hG4bK-p0s3rxrcd7ya;rport From: "2219" ;tag=r7jeyhhxyv To: "2219" Call-ID: 50250e9b9684-8d3g9c7o4hy2 CSeq: 42 REGISTER Max-Forwards: 70 Contact: ;reg-id=1;q=1.0;+sip.instance="";audio;mobility="fixed";duplex="full";description="snom300";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO" User-Agent: snom300/8.7.3.10 Allow-Events: dialog X-Real-IP: 192.168.1.106 Supported: path, gruu Authorization: Digest username="2219",realm="asterisk",nonce="5d67a532",uri="sip:192.168.1.84",response="7e53bd4cac283cf396ca75143abd28c8",algorithm=MD5 Content-Length: 0 <-------------> [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: Header 0 [ 33]: REGISTER sip:192.168.1.84 SIP/2.0 [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: Header 1 [ 69]: Via: SIP/2.0/UDP 192.168.1.106:2048;branch=z9hG4bK-p0s3rxrcd7ya;rport [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: Header 2 [ 51]: From: "2219" ;tag=r7jeyhhxyv [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: Header 3 [ 34]: To: "2219" [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: Header 4 [ 34]: Call-ID: 50250e9b9684-8d3g9c7o4hy2 [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: Header 5 [ 17]: CSeq: 42 REGISTER [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: Header 7 [306]: Contact: ;reg-id=1;q=1.0;+sip.instance="";audio;mobility="fixed";duplex="full";description="snom300";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO" [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: Header 8 [ 28]: User-Agent: snom300/8.7.3.10 [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: Header 9 [ 20]: Allow-Events: dialog [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: Header 10 [ 24]: X-Real-IP: 192.168.1.106 [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: Header 11 [ 21]: Supported: path, gruu [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: Header 12 [152]: Authorization: Digest username="2219",realm="asterisk",nonce="5d67a532",uri="sip:192.168.1.84",response="7e53bd4cac283cf396ca75143abd28c8",algorithm=MD5 [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: Header 13 [ 17]: Content-Length: 0 [Aug 10 15:54:46] VERBOSE[17897] chan_sip.c: --- (14 headers 0 lines) --- [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: = Looking for Call ID: 50250e9b9684-8d3g9c7o4hy2 (Checking From) --From tag r7jeyhhxyv --To-tag [Aug 10 15:54:46] DEBUG[17897] netsock2.c: Splitting '192.168.1.84' into... [Aug 10 15:54:46] DEBUG[17897] netsock2.c: ...host '192.168.1.84' and port ''. [Aug 10 15:54:46] DEBUG[17897] netsock2.c: Splitting '192.168.1.84' into... [Aug 10 15:54:46] DEBUG[17897] netsock2.c: ...host '192.168.1.84' and port ''. [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: Initializing initreq for method REGISTER - callid 50250e9b9684-8d3g9c7o4hy2 [Aug 10 15:54:46] DEBUG[17897] netsock2.c: Splitting '192.168.1.106:2048' into... [Aug 10 15:54:46] DEBUG[17897] netsock2.c: ...host '192.168.1.106' and port '2048'. [Aug 10 15:54:46] VERBOSE[17897] chan_sip.c: Sending to 192.168.1.106:2048 (NAT) [Aug 10 15:54:46] DEBUG[17897] netsock2.c: Splitting '192.168.1.84' into... [Aug 10 15:54:46] DEBUG[17897] netsock2.c: ...host '192.168.1.84' and port ''. [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: Store REGISTER's src-IP:port for call routing. [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: Allocating new SIP dialog for 10906e9c6bc01d4766f986307e0576ce@127.0.0.1:5060 - OPTIONS (No RTP) [Aug 10 15:54:46] DEBUG[17897] acl.c: For destination '192.168.1.106', our source address is '192.168.1.84'. [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.1.84:5060 [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: Initializing initreq for method OPTIONS - callid 02ae89357bd0ba237de090e91a5d44aa@192.168.1.84:5060 [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: Header 0 [ 57]: OPTIONS sip:2219@192.168.1.106:2048;line=7g8ggsvj SIP/2.0 [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK2cfcc28a;rport [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: Header 3 [ 59]: From: "asterisk" ;tag=as2b15deec [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: Header 4 [ 47]: To: [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: Header 5 [ 41]: Contact: [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: Header 6 [ 59]: Call-ID: 02ae89357bd0ba237de090e91a5d44aa@192.168.1.84:5060 [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: Header 8 [ 33]: User-Agent: Asterisk PBX 1.8.15.0 [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: Header 9 [ 35]: Date: Fri, 10 Aug 2012 13:54:46 GMT [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 10 15:54:46] VERBOSE[17897] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.106:2048: OPTIONS sip:2219@192.168.1.106:2048;line=7g8ggsvj SIP/2.0 Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK2cfcc28a;rport Max-Forwards: 70 From: "asterisk" ;tag=as2b15deec To: Contact: Call-ID: 02ae89357bd0ba237de090e91a5d44aa@192.168.1.84:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.15.0 Date: Fri, 10 Aug 2012 13:54:46 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #429 [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.1.106:2048 [Aug 10 15:54:46] VERBOSE[17897] chan_sip.c: <--- Transmitting (NAT) to 192.168.1.106:2048 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.106:2048;branch=z9hG4bK-p0s3rxrcd7ya;received=192.168.1.106;rport=2048 From: "2219" ;tag=r7jeyhhxyv To: "2219" ;tag=as0317aa01 Call-ID: 50250e9b9684-8d3g9c7o4hy2 CSeq: 42 REGISTER Server: Asterisk PBX 1.8.15.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Expires: 120 Contact: ;expires=120 Date: Fri, 10 Aug 2012 13:54:46 GMT Content-Length: 0 <------------> [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.1.106:2048 [Aug 10 15:54:46] VERBOSE[17897] chan_sip.c: Scheduling destruction of SIP dialog '50250e9b9684-8d3g9c7o4hy2' in 32000 ms (Method: REGISTER) [Aug 10 15:54:46] DEBUG[17874] devicestate.c: No provider found, checking channel drivers for SIP - 2219 [Aug 10 15:54:46] DEBUG[17874] chan_sip.c: Checking device state for peer 2219 [Aug 10 15:54:46] DEBUG[17874] devicestate.c: Changing state for SIP/2219 - state 1 (Not in use) [Aug 10 15:54:46] DEBUG[17874] devicestate.c: device 'SIP/2219' state '1' [Aug 10 15:54:46] DEBUG[17907] app_queue.c: Device 'SIP/2219' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Aug 10 15:54:46] VERBOSE[17897] chan_sip.c: <--- SIP read from UDP:192.168.1.106:2048 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK2cfcc28a;rport=5060 From: "asterisk" ;tag=as2b15deec To: ;tag=pod1faajnw Call-ID: 02ae89357bd0ba237de090e91a5d44aa@192.168.1.84:5060 CSeq: 102 OPTIONS Contact: ;reg-id=1 User-Agent: snom300/8.7.3.10 Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: 100rel, replaces, from-change Content-Length: 0 <-------------> [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: Header 1 [ 68]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK2cfcc28a;rport=5060 [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: Header 2 [ 59]: From: "asterisk" ;tag=as2b15deec [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: Header 3 [ 62]: To: ;tag=pod1faajnw [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: Header 4 [ 59]: Call-ID: 02ae89357bd0ba237de090e91a5d44aa@192.168.1.84:5060 [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: Header 6 [ 61]: Contact: ;reg-id=1 [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: Header 7 [ 28]: User-Agent: snom300/8.7.3.10 [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: Header 8 [ 19]: Accept-Language: en [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: Header 10 [ 96]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: Header 11 [ 42]: Allow-Events: talk, hold, refer, call-info [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: Header 12 [ 40]: Supported: 100rel, replaces, from-change [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: Header 13 [ 17]: Content-Length: 0 [Aug 10 15:54:46] VERBOSE[17897] chan_sip.c: --- (14 headers 0 lines) --- [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: = Looking for Call ID: 02ae89357bd0ba237de090e91a5d44aa@192.168.1.84:5060 (Checking To) --From tag as2b15deec --To-tag pod1faajnw [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #429 [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: Stopping retransmission on '02ae89357bd0ba237de090e91a5d44aa@192.168.1.84:5060' of Request 102: Match Found [Aug 10 15:54:46] DEBUG[17897] chan_sip.c: Destroying SIP dialog 02ae89357bd0ba237de090e91a5d44aa@192.168.1.84:5060 [Aug 10 15:54:46] VERBOSE[17897] chan_sip.c: Really destroying SIP dialog '02ae89357bd0ba237de090e91a5d44aa@192.168.1.84:5060' Method: OPTIONS [Aug 10 15:54:47] DEBUG[18238] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 10 15:54:47] WARNING[18238] res_srtp.c: SRTP unprotect failed with: authentication failure 110 [Aug 10 15:54:48] VERBOSE[17897] chan_sip.c: <--- SIP read from UDP:192.168.1.102:2048 ---> BYE sip:2210@192.168.1.84:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.102:2048;branch=z9hG4bK-uhv9js9t63a5;rport From: ;tag=qya7y2dy4z To: "Unknown" ;tag=as497f3fae Call-ID: 542a95eb6183b28264e5cfc965b562ff@192.168.1.84:5060 CSeq: 1 BYE Max-Forwards: 70 Contact: ;reg-id=1 User-Agent: snom300/8.4.32 RTP-RxStat: Total_Rx_Pkts=36395,Rx_Pkts=0,Rx_Pkts_Lost=1,Remote_Rx_Pkts_Lost=7210212 RTP-TxStat: Total_Tx_Pkts=37482,Tx_Pkts=37482,Remote_Tx_Pkts=-1986748224 Content-Length: 0 <-------------> [Aug 10 15:54:48] DEBUG[17897] chan_sip.c: Header 0 [ 38]: BYE sip:2210@192.168.1.84:5060 SIP/2.0 [Aug 10 15:54:48] DEBUG[17897] chan_sip.c: Header 1 [ 69]: Via: SIP/2.0/UDP 192.168.1.102:2048;branch=z9hG4bK-uhv9js9t63a5;rport [Aug 10 15:54:48] DEBUG[17897] chan_sip.c: Header 2 [ 64]: From: ;tag=qya7y2dy4z [Aug 10 15:54:48] DEBUG[17897] chan_sip.c: Header 3 [ 52]: To: "Unknown" ;tag=as497f3fae [Aug 10 15:54:48] DEBUG[17897] chan_sip.c: Header 4 [ 59]: Call-ID: 542a95eb6183b28264e5cfc965b562ff@192.168.1.84:5060 [Aug 10 15:54:48] DEBUG[17897] chan_sip.c: Header 5 [ 11]: CSeq: 1 BYE [Aug 10 15:54:48] DEBUG[17897] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Aug 10 15:54:48] DEBUG[17897] chan_sip.c: Header 7 [ 61]: Contact: ;reg-id=1 [Aug 10 15:54:48] DEBUG[17897] chan_sip.c: Header 8 [ 26]: User-Agent: snom300/8.4.32 [Aug 10 15:54:48] DEBUG[17897] chan_sip.c: Header 9 [ 84]: RTP-RxStat: Total_Rx_Pkts=36395,Rx_Pkts=0,Rx_Pkts_Lost=1,Remote_Rx_Pkts_Lost=7210212 [Aug 10 15:54:48] DEBUG[17897] chan_sip.c: Header 10 [ 72]: RTP-TxStat: Total_Tx_Pkts=37482,Tx_Pkts=37482,Remote_Tx_Pkts=-1986748224 [Aug 10 15:54:48] DEBUG[17897] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 10 15:54:48] VERBOSE[17897] chan_sip.c: --- (12 headers 0 lines) --- [Aug 10 15:54:48] DEBUG[17897] chan_sip.c: = Looking for Call ID: 542a95eb6183b28264e5cfc965b562ff@192.168.1.84:5060 (Checking From) --From tag qya7y2dy4z --To-tag as497f3fae [Aug 10 15:54:48] DEBUG[17897] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [Aug 10 15:54:48] DEBUG[17897] chan_sip.c: Initializing initreq for method BYE - callid 542a95eb6183b28264e5cfc965b562ff@192.168.1.84:5060 [Aug 10 15:54:48] DEBUG[17897] netsock2.c: Splitting '192.168.1.102:2048' into... [Aug 10 15:54:48] DEBUG[17897] netsock2.c: ...host '192.168.1.102' and port '2048'. [Aug 10 15:54:48] VERBOSE[17897] chan_sip.c: Sending to 192.168.1.102:2048 (NAT) [Aug 10 15:54:48] DEBUG[17897] chan_sip.c: Setting SIP_ALREADYGONE on dialog 542a95eb6183b28264e5cfc965b562ff@192.168.1.84:5060 [Aug 10 15:54:48] DEBUG[17897] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x1c51c018' [Aug 10 15:54:48] VERBOSE[17897] chan_sip.c: Scheduling destruction of SIP dialog '542a95eb6183b28264e5cfc965b562ff@192.168.1.84:5060' in 6400 ms (Method: BYE) [Aug 10 15:54:48] DEBUG[17897] chan_sip.c: Received bye, issuing owner hangup [Aug 10 15:54:48] VERBOSE[17897] chan_sip.c: <--- Transmitting (NAT) to 192.168.1.102:2048 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.102:2048;branch=z9hG4bK-uhv9js9t63a5;received=192.168.1.102;rport=2048 From: ;tag=qya7y2dy4z To: "Unknown" ;tag=as497f3fae Call-ID: 542a95eb6183b28264e5cfc965b562ff@192.168.1.84:5060 CSeq: 1 BYE Server: Asterisk PBX 1.8.15.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> [Aug 10 15:54:48] DEBUG[17897] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.1.102:2048 [Aug 10 15:54:48] DEBUG[18238] channel.c: Didn't get a frame from channel: SIP/2212-0000000c [Aug 10 15:54:48] DEBUG[18238] res_rtp_asterisk.c: Setting the marker bit due to a source update [Aug 10 15:54:48] DEBUG[18238] channel.c: Bridge stops bridging channels SIP/2210-0000000b and SIP/2212-0000000c [Aug 10 15:54:48] DEBUG[18238] channel.c: Hanging up channel 'SIP/2212-0000000c' [Aug 10 15:54:48] DEBUG[18238] chan_sip.c: Hangup call SIP/2212-0000000c, SIP callid 542a95eb6183b28264e5cfc965b562ff@192.168.1.84:5060 [Aug 10 15:54:48] DEBUG[18238] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x1c51c018' [Aug 10 15:54:48] DEBUG[18238] app_dial.c: Exiting with DIALSTATUS=ANSWER. [Aug 10 15:54:48] DEBUG[18238] pbx.c: Spawn extension (test,2212,1) exited non-zero on 'SIP/2210-0000000b' [Aug 10 15:54:48] VERBOSE[18238] pbx.c: == Spawn extension (test, 2212, 1) exited non-zero on 'SIP/2210-0000000b' [Aug 10 15:54:48] DEBUG[18238] channel.c: Soft-Hanging up channel 'SIP/2210-0000000b' [Aug 10 15:54:48] DEBUG[18238] channel.c: Hanging up channel 'SIP/2210-0000000b' [Aug 10 15:54:48] DEBUG[18238] chan_sip.c: Hangup call SIP/2210-0000000b, SIP callid 3c2672a5cbc0-o7llh150d2xu [Aug 10 15:54:48] DEBUG[18238] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x1c613e28' [Aug 10 15:54:48] VERBOSE[18238] chan_sip.c: Scheduling destruction of SIP dialog '3c2672a5cbc0-o7llh150d2xu' in 6400 ms (Method: ACK) [Aug 10 15:54:48] DEBUG[18238] chan_sip.c: Strict routing enforced for session 3c2672a5cbc0-o7llh150d2xu [Aug 10 15:54:48] VERBOSE[18238] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 10 15:54:48] DEBUG[18238] netsock2.c: Splitting '192.168.2.210:2048' into... [Aug 10 15:54:48] DEBUG[18238] netsock2.c: ...host '192.168.2.210' and port '2048'. [Aug 10 15:54:48] VERBOSE[18238] chan_sip.c: set_destination: set destination to 192.168.2.210:2048 [Aug 10 15:54:48] VERBOSE[18238] chan_sip.c: Reliably Transmitting (NAT) to 192.168.2.210:2048: BYE sip:2210@192.168.2.210:2048 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK1bfa477a;rport Max-Forwards: 70 From: ;tag=as77bb01b2 To: "2210" ;tag=llqi1mlu1l Call-ID: 3c2672a5cbc0-o7llh150d2xu CSeq: 102 BYE User-Agent: Asterisk PBX 1.8.15.0 Proxy-Authorization: Digest username="2210", realm="asterisk", algorithm=MD5, uri="sip:192.168.1.84", nonce="", response="01a8e78785f2ba2e205612fc4dbb1437" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [Aug 10 15:54:48] DEBUG[18238] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #435 [Aug 10 15:54:48] DEBUG[18238] chan_sip.c: Trying to put 'BYE sip:221' onto UDP socket destined for 192.168.2.210:2048 [Aug 10 15:54:48] DEBUG[17874] devicestate.c: No provider found, checking channel drivers for SIP - 2212 [Aug 10 15:54:48] DEBUG[17874] chan_sip.c: Checking device state for peer 2212 [Aug 10 15:54:48] DEBUG[17874] devicestate.c: Changing state for SIP/2212 - state 1 (Not in use) [Aug 10 15:54:48] DEBUG[17874] devicestate.c: device 'SIP/2212' state '1' [Aug 10 15:54:48] DEBUG[17874] devicestate.c: No provider found, checking channel drivers for SIP - 2210 [Aug 10 15:54:48] DEBUG[17874] chan_sip.c: Checking device state for peer 2210 [Aug 10 15:54:48] DEBUG[17874] devicestate.c: Changing state for SIP/2210 - state 1 (Not in use) [Aug 10 15:54:48] DEBUG[17874] devicestate.c: device 'SIP/2210' state '1' [Aug 10 15:54:48] DEBUG[17907] app_queue.c: Device 'SIP/2212' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Aug 10 15:54:48] DEBUG[17907] app_queue.c: Device 'SIP/2210' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Aug 10 15:54:48] VERBOSE[17897] chan_sip.c: <--- SIP read from UDP:192.168.2.210:2048 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK1bfa477a;rport=5060 From: ;tag=as77bb01b2 To: "2210" ;tag=llqi1mlu1l Call-ID: 3c2672a5cbc0-o7llh150d2xu CSeq: 102 BYE Contact: ;reg-id=1 User-Agent: snom300/8.4.32 RTP-RxStat: Total_Rx_Pkts=37195,Rx_Pkts=0,Rx_Pkts_Lost=1,Remote_Rx_Pkts_Lost=0 RTP-TxStat: Total_Tx_Pkts=37201,Tx_Pkts=1086,Remote_Tx_Pkts=37117 Content-Length: 0 <-------------> [Aug 10 15:54:48] DEBUG[17897] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 10 15:54:48] DEBUG[17897] chan_sip.c: Header 1 [ 68]: Via: SIP/2.0/UDP 192.168.1.84:5060;branch=z9hG4bK1bfa477a;rport=5060 [Aug 10 15:54:48] DEBUG[17897] chan_sip.c: Header 2 [ 44]: From: ;tag=as77bb01b2 [Aug 10 15:54:48] DEBUG[17897] chan_sip.c: Header 3 [ 49]: To: "2210" ;tag=llqi1mlu1l [Aug 10 15:54:48] DEBUG[17897] chan_sip.c: Header 4 [ 34]: Call-ID: 3c2672a5cbc0-o7llh150d2xu [Aug 10 15:54:48] DEBUG[17897] chan_sip.c: Header 5 [ 13]: CSeq: 102 BYE [Aug 10 15:54:48] DEBUG[17897] chan_sip.c: Header 6 [ 47]: Contact: ;reg-id=1 [Aug 10 15:54:48] DEBUG[17897] chan_sip.c: Header 7 [ 26]: User-Agent: snom300/8.4.32 [Aug 10 15:54:48] DEBUG[17897] chan_sip.c: Header 8 [ 78]: RTP-RxStat: Total_Rx_Pkts=37195,Rx_Pkts=0,Rx_Pkts_Lost=1,Remote_Rx_Pkts_Lost=0 [Aug 10 15:54:48] DEBUG[17897] chan_sip.c: Header 9 [ 65]: RTP-TxStat: Total_Tx_Pkts=37201,Tx_Pkts=1086,Remote_Tx_Pkts=37117 [Aug 10 15:54:48] DEBUG[17897] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 10 15:54:48] VERBOSE[17897] chan_sip.c: --- (11 headers 0 lines) --- [Aug 10 15:54:48] DEBUG[17897] chan_sip.c: = Looking for Call ID: 3c2672a5cbc0-o7llh150d2xu (Checking To) --From tag as77bb01b2 --To-tag llqi1mlu1l [Aug 10 15:54:48] DEBUG[17897] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #435 [Aug 10 15:54:48] DEBUG[17897] chan_sip.c: Stopping retransmission on '3c2672a5cbc0-o7llh150d2xu' of Request 102: Match Found [Aug 10 15:54:48] VERBOSE[17897] chan_sip.c: SIP Response message for INCOMING dialog BYE arrived [Aug 10 15:54:48] DEBUG[17897] chan_sip.c: Destroying SIP dialog 3c2672a5cbc0-o7llh150d2xu [Aug 10 15:54:48] VERBOSE[17897] chan_sip.c: Really destroying SIP dialog '3c2672a5cbc0-o7llh150d2xu' Method: ACK [Aug 10 15:54:48] DEBUG[17897] rtp_engine.c: Destroyed RTP instance '0x1c613e28'