~ # cat /var/log/asterisk/myfull [Jun 29 18:10:00] VERBOSE[18564] config.c: == Parsing '/etc/asterisk/logger.conf': [Jun 29 18:10:00] DEBUG[18564] config.c: Parsing /etc/asterisk/logger.conf [Jun 29 18:10:00] VERBOSE[18564] config.c: == Found [Jun 29 18:10:00] VERBOSE[18564] logger.c: Asterisk Queue Logger restarted [Jun 29 18:10:08] DEBUG[18564] chan_sip.c: Asked to create a SIP channel with formats: 0x40 (slin) [Jun 29 18:10:08] DEBUG[18564] chan_sip.c: Allocating new SIP dialog for 357f391167d6768215ee78274b09006d@127.0.0.1 - INVITE (No RTP) [Jun 29 18:10:08] DEBUG[18564] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x962cc30' [Jun 29 18:10:08] DEBUG[18564] res_rtp_asterisk.c: Allocated port 17474 for RTP instance '0x962cc30' [Jun 29 18:10:08] DEBUG[18564] rtp_engine.c: RTP instance '0x962cc30' is setup and ready to go [Jun 29 18:10:08] DEBUG[18564] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x962cc30' [Jun 29 18:10:08] DEBUG[18564] chan_sip.c: Setting NAT on RTP to Off [Jun 29 18:10:08] DEBUG[18564] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Jun 29 18:10:08] DEBUG[18564] acl.c: Found IP address for this socket [Jun 29 18:10:08] DEBUG[18564] chan_sip.c: Setting SIP_TRANSPORT_TLS with address 192.168.20.254:5061 [Jun 29 18:10:08] DEBUG[18564] chan_sip.c: *** Our native formats are 0x4 (ulaw) [Jun 29 18:10:08] DEBUG[18564] chan_sip.c: *** Joint capabilities are 0x0 (nothing) [Jun 29 18:10:08] DEBUG[18564] chan_sip.c: *** Our capabilities are 0xc (ulaw|alaw) [Jun 29 18:10:08] DEBUG[18564] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [Jun 29 18:10:08] DEBUG[18564] chan_sip.c: *** Our preferred formats from the incoming channel are 0x40 (slin) [Jun 29 18:10:08] DEBUG[18564] chan_sip.c: This channel will not be able to handle video. [Jun 29 18:10:08] DEBUG[18564] chan_sip.c: Outgoing Call for 123111113 [Jun 29 18:10:08] DEBUG[18564] chan_sip.c: Updating call counter for outgoing call [Jun 29 18:10:08] DEBUG[18564] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: False Text flag: False [Jun 29 18:10:08] DEBUG[18564] chan_sip.c: ** Our prefcodec: 0x40 (slin) [Jun 29 18:10:08] VERBOSE[18564] chan_sip.c: Audio is at 192.168.20.254 port 17474 [Jun 29 18:10:08] DEBUG[18564] sip/sdp_crypto.c: local_key64 3hadqS245ahz0DUahO3Vvl8SStccliDlaoXt7PoM len 40 [Jun 29 18:10:08] VERBOSE[18564] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Jun 29 18:10:08] VERBOSE[18564] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Jun 29 18:10:08] VERBOSE[18564] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Jun 29 18:10:08] DEBUG[18564] chan_sip.c: -- Done with adding codecs to SDP [Jun 29 18:10:08] DEBUG[18564] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Jun 29 18:10:08] DEBUG[18564] chan_sip.c: Initializing initreq for method INVITE - callid 378153b7405c1a3754d6309121c46961@192.168.20.254 [Jun 29 18:10:08] DEBUG[18564] chan_sip.c: Header 0 [ 62]: INVITE sip:123111113@192.168.20.106:5062;transport=TLS SIP/2.0 [Jun 29 18:10:08] DEBUG[18564] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/TLS 192.168.20.254:5061;branch=z9hG4bK17cebed0 [Jun 29 18:10:08] DEBUG[18564] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jun 29 18:10:08] DEBUG[18564] chan_sip.c: Header 3 [ 61]: From: "asterisk" ;tag=as43b34054 [Jun 29 18:10:08] DEBUG[18564] chan_sip.c: Header 4 [ 53]: To: [Jun 29 18:10:08] DEBUG[18564] chan_sip.c: Header 5 [ 52]: Contact: [Jun 29 18:10:08] DEBUG[18564] chan_sip.c: Header 6 [ 56]: Call-ID: 378153b7405c1a3754d6309121c46961@192.168.20.254 [Jun 29 18:10:08] DEBUG[18564] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Jun 29 18:10:08] DEBUG[18564] chan_sip.c: Header 8 [ 43]: User-Agent: Asterisk PBX SVN-trunk-r272447M [Jun 29 18:10:08] DEBUG[18564] chan_sip.c: Header 9 [ 35]: Date: Tue, 29 Jun 2010 14:10:08 GMT [Jun 29 18:10:08] DEBUG[18564] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Jun 29 18:10:08] DEBUG[18564] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Jun 29 18:10:08] DEBUG[18564] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Jun 29 18:10:08] VERBOSE[18564] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.20.106:5062: INVITE sip:123111113@192.168.20.106:5062;transport=TLS SIP/2.0 Via: SIP/2.0/TLS 192.168.20.254:5061;branch=z9hG4bK17cebed0 Max-Forwards: 70 From: "asterisk" ;tag=as43b34054 To: Contact: Call-ID: 378153b7405c1a3754d6309121c46961@192.168.20.254 CSeq: 102 INVITE User-Agent: Asterisk PBX SVN-trunk-r272447M Date: Tue, 29 Jun 2010 14:10:08 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 358 v=0 o=root 859566719 859566719 IN IP4 192.168.20.254 s=Asterisk PBX SVN-trunk-r272447M c=IN IP4 192.168.20.254 t=0 0 m=audio 17474 RTP/SAVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:3hadqS245ahz0DUahO3Vvl8SStccliDlaoXt7PoM --- [Jun 29 18:10:08] DEBUG[18564] chan_sip.c: Trying to put 'INVITE sip:' onto TLS socket destined for 192.168.20.106:5062 [Jun 29 18:10:08] DEBUG[18565] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Jun 29 18:10:08] DEBUG[18565] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/TLS 192.168.20.254:5061;branch=z9hG4bK17cebed0 [Jun 29 18:10:08] DEBUG[18565] chan_sip.c: Header 2 [ 61]: From: "asterisk" ;tag=as43b34054 [Jun 29 18:10:08] DEBUG[18565] chan_sip.c: Header 3 [ 53]: To: [Jun 29 18:10:08] DEBUG[18565] chan_sip.c: Header 4 [ 56]: Call-ID: 378153b7405c1a3754d6309121c46961@192.168.20.254 [Jun 29 18:10:08] DEBUG[18565] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Jun 29 18:10:08] DEBUG[18565] chan_sip.c: Header 6 [ 38]: User-Agent: Yealink SIP-T26P 6.50.0.50 [Jun 29 18:10:08] DEBUG[18565] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [Jun 29 18:10:08] DEBUG[18565] chan_sip.c: Header 8 [ 0]: [Jun 29 18:10:08] VERBOSE[18565] chan_sip.c: <--- SIP read from TLS:192.168.20.106:5062 ---> SIP/2.0 100 Trying Via: SIP/2.0/TLS 192.168.20.254:5061;branch=z9hG4bK17cebed0 From: "asterisk" ;tag=as43b34054 To: Call-ID: 378153b7405c1a3754d6309121c46961@192.168.20.254 CSeq: 102 INVITE User-Agent: Yealink SIP-T26P 6.50.0.50 Content-Length: 0 <-------------> [Jun 29 18:10:08] DEBUG[18565] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Jun 29 18:10:08] DEBUG[18565] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/TLS 192.168.20.254:5061;branch=z9hG4bK17cebed0 [Jun 29 18:10:08] DEBUG[18565] chan_sip.c: Header 2 [ 61]: From: "asterisk" ;tag=as43b34054 [Jun 29 18:10:08] DEBUG[18565] chan_sip.c: Header 3 [ 53]: To: [Jun 29 18:10:08] DEBUG[18565] chan_sip.c: Header 4 [ 56]: Call-ID: 378153b7405c1a3754d6309121c46961@192.168.20.254 [Jun 29 18:10:08] DEBUG[18565] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Jun 29 18:10:08] DEBUG[18565] chan_sip.c: Header 6 [ 38]: User-Agent: Yealink SIP-T26P 6.50.0.50 [Jun 29 18:10:08] DEBUG[18565] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [Jun 29 18:10:08] DEBUG[18565] chan_sip.c: Header 8 [ 0]: [Jun 29 18:10:08] VERBOSE[18565] chan_sip.c: --- (8 headers 0 lines) --- [Jun 29 18:10:08] DEBUG[18565] chan_sip.c: SIP response 100 to standard invite [Jun 29 18:10:09] DEBUG[18565] chan_sip.c: Header 0 [ 19]: SIP/2.0 180 Ringing [Jun 29 18:10:09] DEBUG[18565] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/TLS 192.168.20.254:5061;branch=z9hG4bK17cebed0 [Jun 29 18:10:09] DEBUG[18565] chan_sip.c: Header 2 [ 61]: From: "asterisk" ;tag=as43b34054 [Jun 29 18:10:09] DEBUG[18565] chan_sip.c: Header 3 [ 67]: To: ;tag=405122912 [Jun 29 18:10:09] DEBUG[18565] chan_sip.c: Header 4 [ 56]: Call-ID: 378153b7405c1a3754d6309121c46961@192.168.20.254 [Jun 29 18:10:09] DEBUG[18565] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Jun 29 18:10:09] DEBUG[18565] chan_sip.c: Header 6 [ 58]: Contact: [Jun 29 18:10:09] DEBUG[18565] chan_sip.c: Header 7 [ 38]: User-Agent: Yealink SIP-T26P 6.50.0.50 [Jun 29 18:10:09] DEBUG[18565] chan_sip.c: Header 8 [ 51]: Allow-Events: talk,hold,conference,refer,check-sync [Jun 29 18:10:09] DEBUG[18565] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Jun 29 18:10:09] DEBUG[18565] chan_sip.c: Header 10 [ 0]: [Jun 29 18:10:09] VERBOSE[18565] chan_sip.c: <--- SIP read from TLS:192.168.20.106:5062 ---> SIP/2.0 180 Ringing Via: SIP/2.0/TLS 192.168.20.254:5061;branch=z9hG4bK17cebed0 From: "asterisk" ;tag=as43b34054 To: ;tag=405122912 Call-ID: 378153b7405c1a3754d6309121c46961@192.168.20.254 CSeq: 102 INVITE Contact: User-Agent: Yealink SIP-T26P 6.50.0.50 Allow-Events: talk,hold,conference,refer,check-sync Content-Length: 0 <-------------> [Jun 29 18:10:09] DEBUG[18565] chan_sip.c: Header 0 [ 19]: SIP/2.0 180 Ringing [Jun 29 18:10:09] DEBUG[18565] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/TLS 192.168.20.254:5061;branch=z9hG4bK17cebed0 [Jun 29 18:10:09] DEBUG[18565] chan_sip.c: Header 2 [ 61]: From: "asterisk" ;tag=as43b34054 [Jun 29 18:10:09] DEBUG[18565] chan_sip.c: Header 3 [ 67]: To: ;tag=405122912 [Jun 29 18:10:09] DEBUG[18565] chan_sip.c: Header 4 [ 56]: Call-ID: 378153b7405c1a3754d6309121c46961@192.168.20.254 [Jun 29 18:10:09] DEBUG[18565] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Jun 29 18:10:09] DEBUG[18565] chan_sip.c: Header 6 [ 58]: Contact: [Jun 29 18:10:09] DEBUG[18565] chan_sip.c: Header 7 [ 38]: User-Agent: Yealink SIP-T26P 6.50.0.50 [Jun 29 18:10:09] DEBUG[18565] chan_sip.c: Header 8 [ 51]: Allow-Events: talk,hold,conference,refer,check-sync [Jun 29 18:10:09] DEBUG[18565] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Jun 29 18:10:09] DEBUG[18565] chan_sip.c: Header 10 [ 0]: [Jun 29 18:10:09] VERBOSE[18565] chan_sip.c: --- (10 headers 0 lines) --- [Jun 29 18:10:09] DEBUG[18565] chan_sip.c: SIP response 180 to standard invite [Jun 29 18:10:09] DEBUG[18501] devicestate.c: No provider found, checking channel drivers for SIP - 123111113 [Jun 29 18:10:09] DEBUG[18501] chan_sip.c: Checking device state for peer 123111113 [Jun 29 18:10:09] DEBUG[18501] devicestate.c: Changing state for SIP/123111113 - state 1 (Not in use) [Jun 29 18:10:09] DEBUG[18501] devicestate.c: device 'SIP/123111113' state '1' [Jun 29 18:10:09] DEBUG[18509] app_queue.c: Device 'SIP/123111113' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jun 29 18:10:13] DEBUG[18512] chan_sip.c: Allocating new SIP dialog for 3f05c8452f8e9dce073051812fa70404@127.0.0.1 - OPTIONS (No RTP) [Jun 29 18:10:13] DEBUG[18512] acl.c: Found IP address for this socket [Jun 29 18:10:13] DEBUG[18512] chan_sip.c: Setting SIP_TRANSPORT_TLS with address 192.168.20.254:5061 [Jun 29 18:10:13] DEBUG[18512] chan_sip.c: Initializing initreq for method OPTIONS - callid 3b558af17947b1d220120c5e0e5ca6f8@192.168.20.254 [Jun 29 18:10:13] DEBUG[18512] chan_sip.c: Header 0 [ 63]: OPTIONS sip:123111113@192.168.20.106:5062;transport=TLS SIP/2.0 [Jun 29 18:10:13] DEBUG[18512] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/TLS 192.168.20.254:5061;branch=z9hG4bK4e993a1c [Jun 29 18:10:13] DEBUG[18512] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jun 29 18:10:13] DEBUG[18512] chan_sip.c: Header 3 [ 61]: From: "asterisk" ;tag=as45705118 [Jun 29 18:10:13] DEBUG[18512] chan_sip.c: Header 4 [ 53]: To: [Jun 29 18:10:13] DEBUG[18512] chan_sip.c: Header 5 [ 52]: Contact: [Jun 29 18:10:13] DEBUG[18512] chan_sip.c: Header 6 [ 56]: Call-ID: 3b558af17947b1d220120c5e0e5ca6f8@192.168.20.254 [Jun 29 18:10:13] DEBUG[18512] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Jun 29 18:10:13] DEBUG[18512] chan_sip.c: Header 8 [ 43]: User-Agent: Asterisk PBX SVN-trunk-r272447M [Jun 29 18:10:13] DEBUG[18512] chan_sip.c: Header 9 [ 35]: Date: Tue, 29 Jun 2010 14:10:13 GMT [Jun 29 18:10:13] DEBUG[18512] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Jun 29 18:10:13] DEBUG[18512] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Jun 29 18:10:13] VERBOSE[18512] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.20.106:5062: OPTIONS sip:123111113@192.168.20.106:5062;transport=TLS SIP/2.0 Via: SIP/2.0/TLS 192.168.20.254:5061;branch=z9hG4bK4e993a1c Max-Forwards: 70 From: "asterisk" ;tag=as45705118 To: Contact: Call-ID: 3b558af17947b1d220120c5e0e5ca6f8@192.168.20.254 CSeq: 102 OPTIONS User-Agent: Asterisk PBX SVN-trunk-r272447M Date: Tue, 29 Jun 2010 14:10:13 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Jun 29 18:10:13] DEBUG[18512] chan_sip.c: Trying to put 'OPTIONS sip' onto TLS socket destined for 192.168.20.106:5062 [Jun 29 18:10:13] DEBUG[18565] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Jun 29 18:10:13] DEBUG[18565] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/TLS 192.168.20.254:5061;branch=z9hG4bK4e993a1c [Jun 29 18:10:13] DEBUG[18565] chan_sip.c: Header 2 [ 61]: From: "asterisk" ;tag=as45705118 [Jun 29 18:10:13] DEBUG[18565] chan_sip.c: Header 3 [ 67]: To: ;tag=816000978 [Jun 29 18:10:13] DEBUG[18565] chan_sip.c: Header 4 [ 56]: Call-ID: 3b558af17947b1d220120c5e0e5ca6f8@192.168.20.254 [Jun 29 18:10:13] DEBUG[18565] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Jun 29 18:10:13] DEBUG[18565] chan_sip.c: Header 6 [ 38]: User-Agent: Yealink SIP-T26P 6.50.0.50 [Jun 29 18:10:13] DEBUG[18565] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [Jun 29 18:10:13] DEBUG[18565] chan_sip.c: Header 8 [ 0]: [Jun 29 18:10:13] VERBOSE[18565] chan_sip.c: <--- SIP read from TLS:192.168.20.106:5062 ---> SIP/2.0 200 OK Via: SIP/2.0/TLS 192.168.20.254:5061;branch=z9hG4bK4e993a1c From: "asterisk" ;tag=as45705118 To: ;tag=816000978 Call-ID: 3b558af17947b1d220120c5e0e5ca6f8@192.168.20.254 CSeq: 102 OPTIONS User-Agent: Yealink SIP-T26P 6.50.0.50 Content-Length: 0 <-------------> [Jun 29 18:10:13] DEBUG[18565] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Jun 29 18:10:13] DEBUG[18565] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/TLS 192.168.20.254:5061;branch=z9hG4bK4e993a1c [Jun 29 18:10:13] DEBUG[18565] chan_sip.c: Header 2 [ 61]: From: "asterisk" ;tag=as45705118 [Jun 29 18:10:13] DEBUG[18565] chan_sip.c: Header 3 [ 67]: To: ;tag=816000978 [Jun 29 18:10:13] DEBUG[18565] chan_sip.c: Header 4 [ 56]: Call-ID: 3b558af17947b1d220120c5e0e5ca6f8@192.168.20.254 [Jun 29 18:10:13] DEBUG[18565] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Jun 29 18:10:13] DEBUG[18565] chan_sip.c: Header 6 [ 38]: User-Agent: Yealink SIP-T26P 6.50.0.50 [Jun 29 18:10:13] DEBUG[18565] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [Jun 29 18:10:13] DEBUG[18565] chan_sip.c: Header 8 [ 0]: [Jun 29 18:10:13] VERBOSE[18565] chan_sip.c: --- (8 headers 0 lines) --- [Jun 29 18:10:13] DEBUG[18565] chan_sip.c: Header 0 [ 0]: [Jun 29 18:10:13] DEBUG[18565] chan_sip.c: Body 0 [ 0]: [Jun 29 18:10:13] VERBOSE[18565] chan_sip.c: <--- SIP read from TLS:192.168.20.106:5062 ---> <-------------> [Jun 29 18:10:13] DEBUG[18565] chan_sip.c: Header 0 [ 0]: [Jun 29 18:10:13] DEBUG[18565] chan_sip.c: Body 0 [ 0]: [Jun 29 18:10:14] DEBUG[18512] chan_sip.c: Destroying SIP dialog 3b558af17947b1d220120c5e0e5ca6f8@192.168.20.254 [Jun 29 18:10:14] VERBOSE[18512] chan_sip.c: Really destroying SIP dialog '3b558af17947b1d220120c5e0e5ca6f8@192.168.20.254' Method: OPTIONS [Jun 29 18:10:15] DEBUG[18565] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Jun 29 18:10:15] DEBUG[18565] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/TLS 192.168.20.254:5061;branch=z9hG4bK17cebed0 [Jun 29 18:10:15] DEBUG[18565] chan_sip.c: Header 2 [ 61]: From: "asterisk" ;tag=as43b34054 [Jun 29 18:10:15] DEBUG[18565] chan_sip.c: Header 3 [ 67]: To: ;tag=405122912 [Jun 29 18:10:15] DEBUG[18565] chan_sip.c: Header 4 [ 56]: Call-ID: 378153b7405c1a3754d6309121c46961@192.168.20.254 [Jun 29 18:10:15] DEBUG[18565] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Jun 29 18:10:15] DEBUG[18565] chan_sip.c: Header 6 [ 58]: Contact: [Jun 29 18:10:15] DEBUG[18565] chan_sip.c: Header 7 [ 29]: Content-Type: application/sdp [Jun 29 18:10:15] DEBUG[18565] chan_sip.c: Header 8 [115]: Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE [Jun 29 18:10:15] DEBUG[18565] chan_sip.c: Header 9 [ 38]: User-Agent: Yealink SIP-T26P 6.50.0.50 [Jun 29 18:10:15] DEBUG[18565] chan_sip.c: Header 10 [ 19]: Content-Length: 300 [Jun 29 18:10:15] DEBUG[18565] chan_sip.c: Header 11 [ 0]: [Jun 29 18:10:15] VERBOSE[18565] chan_sip.c: <--- SIP read from TLS:192.168.20.106:5062 ---> SIP/2.0 200 OK Via: SIP/2.0/TLS 192.168.20.254:5061;branch=z9hG4bK17cebed0 From: "asterisk" ;tag=as43b34054 To: ;tag=405122912 Call-ID: 378153b7405c1a3754d6309121c46961@192.168.20.254 CSeq: 102 INVITE Contact: Content-Type: application/sdp Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE User-Agent: Yealink SIP-T26P 6.50.0.50 Content-Length: 300 v=0 o=- 20001 20001 IN IP4 192.168.20.106 s=SDP data c=IN IP4 192.168.20.106 t=0 0 m=audio 11780 RTP/SAVP 0 101 a=rtpmap:0 PCMU/8000 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:MWI1OTFlNmM2Y2M4YWU1MTVhOTQ4ZTQ5NWEzNjBl a=sendrecv a=ptime:20 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 <-------------> [Jun 29 18:10:15] DEBUG[18565] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Jun 29 18:10:15] DEBUG[18565] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/TLS 192.168.20.254:5061;branch=z9hG4bK17cebed0 [Jun 29 18:10:15] DEBUG[18565] chan_sip.c: Header 2 [ 61]: From: "asterisk" ;tag=as43b34054 [Jun 29 18:10:15] DEBUG[18565] chan_sip.c: Header 3 [ 67]: To: ;tag=405122912 [Jun 29 18:10:15] DEBUG[18565] chan_sip.c: Header 4 [ 56]: Call-ID: 378153b7405c1a3754d6309121c46961@192.168.20.254 [Jun 29 18:10:15] DEBUG[18565] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Jun 29 18:10:15] DEBUG[18565] chan_sip.c: Header 6 [ 58]: Contact: [Jun 29 18:10:15] DEBUG[18565] chan_sip.c: Header 7 [ 29]: Content-Type: application/sdp [Jun 29 18:10:15] DEBUG[18565] chan_sip.c: Header 8 [115]: Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE [Jun 29 18:10:15] DEBUG[18565] chan_sip.c: Header 9 [ 38]: User-Agent: Yealink SIP-T26P 6.50.0.50 [Jun 29 18:10:15] DEBUG[18565] chan_sip.c: Header 10 [ 19]: Content-Length: 300 [Jun 29 18:10:15] DEBUG[18565] chan_sip.c: Header 11 [ 0]: [Jun 29 18:10:15] DEBUG[18565] chan_sip.c: Body 0 [ 3]: v=0 [Jun 29 18:10:15] DEBUG[18565] chan_sip.c: Body 1 [ 37]: o=- 20001 20001 IN IP4 192.168.20.106 [Jun 29 18:10:15] DEBUG[18565] chan_sip.c: Body 2 [ 10]: s=SDP data [Jun 29 18:10:15] DEBUG[18565] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.20.106 [Jun 29 18:10:15] DEBUG[18565] chan_sip.c: Body 4 [ 5]: t=0 0 [Jun 29 18:10:15] DEBUG[18565] chan_sip.c: Body 5 [ 28]: m=audio 11780 RTP/SAVP 0 101 [Jun 29 18:10:15] DEBUG[18565] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Jun 29 18:10:15] DEBUG[18565] chan_sip.c: Body 7 [ 82]: a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:MWI1OTFlNmM2Y2M4YWU1MTVhOTQ4ZTQ5NWEzNjBl [Jun 29 18:10:15] DEBUG[18565] chan_sip.c: Body 8 [ 10]: a=sendrecv [Jun 29 18:10:15] DEBUG[18565] chan_sip.c: Body 9 [ 10]: a=ptime:20 [Jun 29 18:10:15] DEBUG[18565] chan_sip.c: Body 10 [ 15]: a=fmtp:101 0-15 [Jun 29 18:10:15] DEBUG[18565] chan_sip.c: Body 11 [ 33]: a=rtpmap:101 telephone-event/8000 [Jun 29 18:10:15] VERBOSE[18565] chan_sip.c: --- (11 headers 12 lines) --- [Jun 29 18:10:15] DEBUG[18565] chan_sip.c: SIP response 200 to standard invite [Jun 29 18:10:15] DEBUG[18565] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Jun 29 18:10:15] DEBUG[18565] chan_sip.c: Processing session-level SDP o=- 20001 20001 IN IP4 192.168.20.106... UNSUPPORTED. [Jun 29 18:10:15] DEBUG[18565] chan_sip.c: Processing session-level SDP s=SDP data... UNSUPPORTED. [Jun 29 18:10:15] DEBUG[18565] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.20.106... OK. [Jun 29 18:10:15] DEBUG[18565] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Jun 29 18:10:15] VERBOSE[18565] chan_sip.c: Found RTP audio format 0 [Jun 29 18:10:15] DEBUG[18565] rtp_engine.c: Setting payload 0 based on m type on 0xb12ec784 [Jun 29 18:10:15] VERBOSE[18565] chan_sip.c: Found RTP audio format 101 [Jun 29 18:10:15] DEBUG[18565] rtp_engine.c: Setting payload 101 based on m type on 0xb12ec784 [Jun 29 18:10:15] VERBOSE[18565] chan_sip.c: Found audio description format PCMU for ID 0 [Jun 29 18:10:15] DEBUG[18565] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Jun 29 18:10:15] DEBUG[18565] sip/sdp_crypto.c: SRTP policy activated [Jun 29 18:10:15] DEBUG[18565] chan_sip.c: Processing media-level (audio) SDP a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:MWI1OTFlNmM2Y2M4YWU1MTVhOTQ4ZTQ5NWEzNjBl... OK. [Jun 29 18:10:15] DEBUG[18565] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Jun 29 18:10:15] DEBUG[18565] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Jun 29 18:10:15] DEBUG[18565] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED. [Jun 29 18:10:15] VERBOSE[18565] chan_sip.c: Found audio description format telephone-event for ID 101 [Jun 29 18:10:15] DEBUG[18565] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Jun 29 18:10:15] DEBUG[18565] rtp_engine.c: Incorporating payload 0 on 0xb12ec784 [Jun 29 18:10:15] DEBUG[18565] rtp_engine.c: Incorporating payload 101 on 0xb12ec784 [Jun 29 18:10:15] VERBOSE[18565] chan_sip.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) [Jun 29 18:10:15] VERBOSE[18565] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Jun 29 18:10:15] DEBUG[18565] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x962cc30' [Jun 29 18:10:15] VERBOSE[18565] chan_sip.c: Peer audio RTP is at port 192.168.20.106:11780 [Jun 29 18:10:15] DEBUG[18565] rtp_engine.c: Copying payload 0 from 0xb12ec784 to 0x962cc80 [Jun 29 18:10:15] DEBUG[18565] rtp_engine.c: Copying payload 101 from 0xb12ec784 to 0x962cc80 [Jun 29 18:10:15] DEBUG[18565] chan_sip.c: We're settling with these formats: 0x4 (ulaw) [Jun 29 18:10:15] DEBUG[18565] chan_sip.c: We have an owner, now see if we need to change this call [Jun 29 18:10:15] DEBUG[18565] chan_sip.c: Updating call counter for outgoing call [Jun 29 18:10:15] DEBUG[18565] chan_sip.c: build_route: Contact hop: [Jun 29 18:10:15] VERBOSE[18565] chan_sip.c: list_route: hop: [Jun 29 18:10:15] DEBUG[18565] chan_sip.c: Strict routing enforced for session 378153b7405c1a3754d6309121c46961@192.168.20.254 [Jun 29 18:10:15] VERBOSE[18565] chan_sip.c: set_destination: Parsing for address/port to send to [Jun 29 18:10:15] VERBOSE[18565] chan_sip.c: set_destination: set destination to 192.168.20.106, port 5062 [Jun 29 18:10:15] VERBOSE[18565] chan_sip.c: Transmitting (no NAT) to 192.168.20.106:5062: ACK sip:123111113@192.168.20.106:5062;transport=TLS SIP/2.0 Via: SIP/2.0/TLS 192.168.20.254:5061;branch=z9hG4bK01edcb7f Max-Forwards: 70 From: "asterisk" ;tag=as43b34054 To: ;tag=405122912 Contact: Call-ID: 378153b7405c1a3754d6309121c46961@192.168.20.254 CSeq: 102 ACK User-Agent: Asterisk PBX SVN-trunk-r272447M Content-Length: 0 --- [Jun 29 18:10:15] DEBUG[18565] chan_sip.c: Trying to put 'ACK sip:123' onto TLS socket destined for 192.168.20.106:5062 [Jun 29 18:10:15] DEBUG[18501] devicestate.c: No provider found, checking channel drivers for SIP - 123111113 [Jun 29 18:10:15] DEBUG[18501] chan_sip.c: Checking device state for peer 123111113 [Jun 29 18:10:15] DEBUG[18501] devicestate.c: Changing state for SIP/123111113 - state 1 (Not in use) [Jun 29 18:10:15] DEBUG[18501] devicestate.c: device 'SIP/123111113' state '1' [Jun 29 18:10:15] DEBUG[18509] app_queue.c: Device 'SIP/123111113' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jun 29 18:10:15] VERBOSE[18566] pbx.c: -- Launching echo() on SIP/123111113-00000002 [Jun 29 18:10:15] DEBUG[18566] res_srtp.c: SRTP unprotect: authentication failure [Jun 29 18:10:15] WARNING[18566] res_rtp_asterisk.c: RTP Read error: Success. Hanging up. [Jun 29 18:10:15] DEBUG[18566] channel.c: Hanging up channel 'SIP/123111113-00000002' [Jun 29 18:10:15] DEBUG[18566] chan_sip.c: Hangup call SIP/123111113-00000002, SIP callid 378153b7405c1a3754d6309121c46961@192.168.20.254 [Jun 29 18:10:15] DEBUG[18566] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x962cc30' [Jun 29 18:10:15] VERBOSE[18566] chan_sip.c: Scheduling destruction of SIP dialog '378153b7405c1a3754d6309121c46961@192.168.20.254' in 7936 ms (Method: INVITE) [Jun 29 18:10:15] DEBUG[18566] chan_sip.c: Strict routing enforced for session 378153b7405c1a3754d6309121c46961@192.168.20.254 [Jun 29 18:10:15] VERBOSE[18566] chan_sip.c: set_destination: Parsing for address/port to send to [Jun 29 18:10:15] VERBOSE[18566] chan_sip.c: set_destination: set destination to 192.168.20.106, port 5062 [Jun 29 18:10:15] VERBOSE[18566] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.20.106:5062: BYE sip:123111113@192.168.20.106:5062;transport=TLS SIP/2.0 Via: SIP/2.0/TLS 192.168.20.254:5061;branch=z9hG4bK01f2be07 Max-Forwards: 70 From: "asterisk" ;tag=as43b34054 To: ;tag=405122912 Call-ID: 378153b7405c1a3754d6309121c46961@192.168.20.254 CSeq: 103 BYE User-Agent: Asterisk PBX SVN-trunk-r272447M X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [Jun 29 18:10:15] DEBUG[18566] chan_sip.c: Trying to put 'BYE sip:123' onto TLS socket destined for 192.168.20.106:5062 [Jun 29 18:10:15] DEBUG[18501] devicestate.c: No provider found, checking channel drivers for SIP - 123111113 [Jun 29 18:10:15] DEBUG[18501] chan_sip.c: Checking device state for peer 123111113 [Jun 29 18:10:15] DEBUG[18501] devicestate.c: Changing state for SIP/123111113 - state 1 (Not in use) [Jun 29 18:10:15] DEBUG[18501] devicestate.c: device 'SIP/123111113' state '1' [Jun 29 18:10:15] DEBUG[18509] app_queue.c: Device 'SIP/123111113' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jun 29 18:10:15] DEBUG[18565] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Jun 29 18:10:15] DEBUG[18565] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/TLS 192.168.20.254:5061;branch=z9hG4bK01f2be07 [Jun 29 18:10:15] DEBUG[18565] chan_sip.c: Header 2 [ 61]: From: "asterisk" ;tag=as43b34054 [Jun 29 18:10:15] DEBUG[18565] chan_sip.c: Header 3 [ 67]: To: ;tag=405122912 [Jun 29 18:10:15] DEBUG[18565] chan_sip.c: Header 4 [ 56]: Call-ID: 378153b7405c1a3754d6309121c46961@192.168.20.254 [Jun 29 18:10:15] DEBUG[18565] chan_sip.c: Header 5 [ 13]: CSeq: 103 BYE [Jun 29 18:10:15] DEBUG[18565] chan_sip.c: Header 6 [ 38]: User-Agent: Yealink SIP-T26P 6.50.0.50 [Jun 29 18:10:15] DEBUG[18565] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [Jun 29 18:10:15] DEBUG[18565] chan_sip.c: Header 8 [ 0]: [Jun 29 18:10:15] VERBOSE[18565] chan_sip.c: <--- SIP read from TLS:192.168.20.106:5062 ---> SIP/2.0 200 OK Via: SIP/2.0/TLS 192.168.20.254:5061;branch=z9hG4bK01f2be07 From: "asterisk" ;tag=as43b34054 To: ;tag=405122912 Call-ID: 378153b7405c1a3754d6309121c46961@192.168.20.254 CSeq: 103 BYE User-Agent: Yealink SIP-T26P 6.50.0.50 Content-Length: 0 <-------------> [Jun 29 18:10:15] DEBUG[18565] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Jun 29 18:10:15] DEBUG[18565] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/TLS 192.168.20.254:5061;branch=z9hG4bK01f2be07 [Jun 29 18:10:15] DEBUG[18565] chan_sip.c: Header 2 [ 61]: From: "asterisk" ;tag=as43b34054 [Jun 29 18:10:15] DEBUG[18565] chan_sip.c: Header 3 [ 67]: To: ;tag=405122912 [Jun 29 18:10:15] DEBUG[18565] chan_sip.c: Header 4 [ 56]: Call-ID: 378153b7405c1a3754d6309121c46961@192.168.20.254 [Jun 29 18:10:15] DEBUG[18565] chan_sip.c: Header 5 [ 13]: CSeq: 103 BYE [Jun 29 18:10:15] DEBUG[18565] chan_sip.c: Header 6 [ 38]: User-Agent: Yealink SIP-T26P 6.50.0.50 [Jun 29 18:10:15] DEBUG[18565] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [Jun 29 18:10:15] DEBUG[18565] chan_sip.c: Header 8 [ 0]: [Jun 29 18:10:15] VERBOSE[18565] chan_sip.c: --- (8 headers 0 lines) --- [Jun 29 18:10:15] DEBUG[18512] chan_sip.c: Destroying SIP dialog 378153b7405c1a3754d6309121c46961@192.168.20.254 [Jun 29 18:10:15] VERBOSE[18512] chan_sip.c: Really destroying SIP dialog '378153b7405c1a3754d6309121c46961@192.168.20.254' Method: INVITE [Jun 29 18:10:15] DEBUG[18512] rtp_engine.c: Destroyed RTP instance '0x962cc30' [Jun 29 18:10:33] VERBOSE[18564] asterisk.c: -- Remote UNIX connection disconnected [Jun 29 18:10:45] DEBUG[18565] chan_sip.c: Header 0 [ 0]: [Jun 29 18:10:45] DEBUG[18565] chan_sip.c: Body 0 [ 0]: [Jun 29 18:10:45] VERBOSE[18565] chan_sip.c: <--- SIP read from TLS:192.168.20.106:5062 ---> <-------------> [Jun 29 18:10:45] DEBUG[18565] chan_sip.c: Header 0 [ 0]: [Jun 29 18:10:45] DEBUG[18565] chan_sip.c: Body 0 [ 0]: