[Jul 6 16:52:06] VERBOSE[29082] config.c: == Parsing '/etc/asterisk/logger.conf': [Jul 6 16:52:06] DEBUG[29082] config.c: Parsing /etc/asterisk/logger.conf [Jul 6 16:52:06] VERBOSE[29082] config.c: == Found [Jul 6 16:52:06] VERBOSE[29082] logger.c: Asterisk Event Logger restarted [Jul 6 16:52:06] VERBOSE[29082] logger.c: Asterisk Queue Logger restarted [Jul 6 16:52:18] DEBUG[29058] chan_sip.c: Allocating new SIP dialog for 20a12c14287162ce311416692d8abc0c@127.0.0.1 - OPTIONS (No RTP) [Jul 6 16:52:18] DEBUG[29058] acl.c: Found IP address for this socket [Jul 6 16:52:18] DEBUG[29058] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 10.10.6.46:5060 [Jul 6 16:52:18] DEBUG[29058] chan_sip.c: Initializing initreq for method OPTIONS - callid 654bc7431266f54e7b4464391e237c3e@10.10.6.46 [Jul 6 16:52:18] DEBUG[29058] chan_sip.c: Header 0 [ 31]: OPTIONS sip:10.10.10.16 SIP/2.0 [Jul 6 16:52:18] DEBUG[29058] chan_sip.c: Header 1 [ 61]: Via: SIP/2.0/UDP 10.10.6.46:5060;branch=z9hG4bK6aadd8f1;rport [Jul 6 16:52:18] DEBUG[29058] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jul 6 16:52:18] DEBUG[29058] chan_sip.c: Header 3 [ 57]: From: "asterisk" ;tag=as1544e14e [Jul 6 16:52:18] DEBUG[29058] chan_sip.c: Header 4 [ 21]: To: [Jul 6 16:52:18] DEBUG[29058] chan_sip.c: Header 5 [ 34]: Contact: [Jul 6 16:52:18] DEBUG[29058] chan_sip.c: Header 6 [ 52]: Call-ID: 654bc7431266f54e7b4464391e237c3e@10.10.6.46 [Jul 6 16:52:18] DEBUG[29058] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Jul 6 16:52:18] DEBUG[29058] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.6.2.9 [Jul 6 16:52:18] DEBUG[29058] chan_sip.c: Header 9 [ 35]: Date: Tue, 06 Jul 2010 23:52:18 GMT [Jul 6 16:52:18] DEBUG[29058] chan_sip.c: Header 10 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Jul 6 16:52:18] DEBUG[29058] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Jul 6 16:52:18] DEBUG[29058] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Jul 6 16:52:18] VERBOSE[29058] chan_sip.c: Reliably Transmitting (no NAT) to 10.10.10.16:5060: OPTIONS sip:10.10.10.16 SIP/2.0 Via: SIP/2.0/UDP 10.10.6.46:5060;branch=z9hG4bK6aadd8f1;rport Max-Forwards: 70 From: "asterisk" ;tag=as1544e14e To: Contact: Call-ID: 654bc7431266f54e7b4464391e237c3e@10.10.6.46 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.9 Date: Tue, 06 Jul 2010 23:52:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- [Jul 6 16:52:18] DEBUG[29058] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #18463 [Jul 6 16:52:18] DEBUG[29058] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 10.10.10.16:5060 [Jul 6 16:52:18] VERBOSE[29058] chan_sip.c: <--- SIP read from UDP:10.10.10.16:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.6.46:5060;branch=z9hG4bK6aadd8f1;rport From: "asterisk" ;tag=as1544e14e To: ;tag=hssUA_1181808736-111488116 Call-ID: 654bc7431266f54e7b4464391e237c3e@10.10.6.46 CSeq: 102 OPTIONS User-Agent: ShoreTel_FSS_SIP_UATK ShoreTel 7 Contact: DefaultProfile Allow: INVITE,BYE,CANCEL,ACK,INFO,PRACK,COMET,OPTIONS,SUBSCRIBE,NOTIFY,MESSAGE,REFER,REGISTER,UPDATE Content-Length: 0 <-------------> [Jul 6 16:52:18] DEBUG[29058] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Jul 6 16:52:18] DEBUG[29058] chan_sip.c: Header 1 [ 61]: Via: SIP/2.0/UDP 10.10.6.46:5060;branch=z9hG4bK6aadd8f1;rport [Jul 6 16:52:18] DEBUG[29058] chan_sip.c: Header 2 [ 57]: From: "asterisk" ;tag=as1544e14e [Jul 6 16:52:18] DEBUG[29058] chan_sip.c: Header 3 [ 52]: To: ;tag=hssUA_1181808736-111488116 [Jul 6 16:52:18] DEBUG[29058] chan_sip.c: Header 4 [ 52]: Call-ID: 654bc7431266f54e7b4464391e237c3e@10.10.6.46 [Jul 6 16:52:18] DEBUG[29058] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Jul 6 16:52:18] DEBUG[29058] chan_sip.c: Header 6 [ 44]: User-Agent: ShoreTel_FSS_SIP_UATK ShoreTel 7 [Jul 6 16:52:18] DEBUG[29058] chan_sip.c: Header 7 [ 61]: Contact: DefaultProfile [Jul 6 16:52:18] DEBUG[29058] chan_sip.c: Header 8 [100]: Allow: INVITE,BYE,CANCEL,ACK,INFO,PRACK,COMET,OPTIONS,SUBSCRIBE,NOTIFY,MESSAGE,REFER,REGISTER,UPDATE [Jul 6 16:52:18] DEBUG[29058] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Jul 6 16:52:18] DEBUG[29058] chan_sip.c: Header 10 [ 0]: [Jul 6 16:52:18] VERBOSE[29058] chan_sip.c: --- (10 headers 0 lines) --- [Jul 6 16:52:18] DEBUG[29058] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #18463 [Jul 6 16:52:18] DEBUG[29058] chan_sip.c: Stopping retransmission on '654bc7431266f54e7b4464391e237c3e@10.10.6.46' of Request 102: Match Found [Jul 6 16:52:18] DEBUG[29058] chan_sip.c: Destroying SIP dialog 654bc7431266f54e7b4464391e237c3e@10.10.6.46 [Jul 6 16:52:18] VERBOSE[29058] chan_sip.c: Really destroying SIP dialog '654bc7431266f54e7b4464391e237c3e@10.10.6.46' Method: OPTIONS [Jul 6 16:52:37] DEBUG[5870] manager.c: Manager received command 'Originate' [Jul 6 16:52:37] DEBUG[5870] chan_sip.c: Asked to create a SIP channel with formats: 0x40 (slin) [Jul 6 16:52:37] VERBOSE[5870] netsock.c: == Using SIP RTP CoS mark 5 [Jul 6 16:52:37] DEBUG[5870] chan_sip.c: Allocating new SIP dialog for 6acf1b346c6bbd267671f40d5c935131@127.0.0.1 - INVITE (With RTP) [Jul 6 16:52:37] DEBUG[5870] chan_sip.c: Setting NAT on RTP to Off [Jul 6 16:52:37] DEBUG[5870] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Jul 6 16:52:37] DEBUG[5870] acl.c: Found IP address for this socket [Jul 6 16:52:37] DEBUG[5870] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 10.10.6.46:5060 [Jul 6 16:52:37] DEBUG[5870] chan_sip.c: *** Our native formats are 0x80004 (ulaw|h263) [Jul 6 16:52:37] DEBUG[5870] chan_sip.c: *** Joint capabilities are 0x0 (nothing) [Jul 6 16:52:37] DEBUG[5870] chan_sip.c: *** Our capabilities are 0x8000e (gsm|ulaw|alaw|h263) [Jul 6 16:52:37] DEBUG[5870] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [Jul 6 16:52:37] DEBUG[5870] chan_sip.c: *** Our preferred formats from the incoming channel are 0x40 (slin) [Jul 6 16:52:37] DEBUG[5870] chan_sip.c: This channel will not be able to handle video. [Jul 6 16:52:37] DEBUG[5870] chan_sip.c: Outgoing Call for [Jul 6 16:52:37] DEBUG[5870] chan_sip.c: Updating call counter for outgoing call [Jul 6 16:52:37] DEBUG[5870] chan_sip.c: This call needs video offers, but there's no video support enabled! [Jul 6 16:52:37] DEBUG[5870] chan_sip.c: ** Our capability: 0x8000e (gsm|ulaw|alaw|h263) Video flag: False Text flag: False [Jul 6 16:52:37] DEBUG[5870] chan_sip.c: ** Our prefcodec: 0x40 (slin) [Jul 6 16:52:37] VERBOSE[5870] chan_sip.c: Audio is at 10.10.6.46 port 15326 [Jul 6 16:52:37] VERBOSE[5870] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Jul 6 16:52:37] VERBOSE[5870] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Jul 6 16:52:37] VERBOSE[5870] chan_sip.c: Adding codec 0x2 (gsm) to SDP [Jul 6 16:52:37] VERBOSE[5870] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Jul 6 16:52:37] DEBUG[5870] chan_sip.c: -- Done with adding codecs to SDP [Jul 6 16:52:37] DEBUG[5870] chan_sip.c: Done building SDP. Settling with this capability: 0x8000e (gsm|ulaw|alaw|h263) [Jul 6 16:52:37] DEBUG[5870] chan_sip.c: Initializing initreq for method INVITE - callid 04f8bd1d4d261c8d4fbb3a0507c9c5ab@10.10.6.46 [Jul 6 16:52:37] DEBUG[5870] chan_sip.c: Header 0 [ 35]: INVITE sip:10.10.10.16:5060 SIP/2.0 [Jul 6 16:52:37] DEBUG[5870] chan_sip.c: Header 1 [ 61]: Via: SIP/2.0/UDP 10.10.6.46:5060;branch=z9hG4bK298d4a7a;rport [Jul 6 16:52:37] DEBUG[5870] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jul 6 16:52:37] DEBUG[5870] chan_sip.c: Header 3 [ 57]: From: "asterisk" ;tag=as317a6b87 [Jul 6 16:52:37] DEBUG[5870] chan_sip.c: Header 4 [ 26]: To: [Jul 6 16:52:37] DEBUG[5870] chan_sip.c: Header 5 [ 34]: Contact: [Jul 6 16:52:37] DEBUG[5870] chan_sip.c: Header 6 [ 52]: Call-ID: 04f8bd1d4d261c8d4fbb3a0507c9c5ab@10.10.6.46 [Jul 6 16:52:37] DEBUG[5870] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Jul 6 16:52:37] DEBUG[5870] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.6.2.9 [Jul 6 16:52:37] DEBUG[5870] chan_sip.c: Header 9 [ 35]: Date: Tue, 06 Jul 2010 23:52:37 GMT [Jul 6 16:52:37] DEBUG[5870] chan_sip.c: Header 10 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Jul 6 16:52:37] DEBUG[5870] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Jul 6 16:52:37] DEBUG[5870] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Jul 6 16:52:37] DEBUG[5870] chan_sip.c: Header 13 [ 19]: Content-Length: 277 [Jul 6 16:52:37] DEBUG[5870] chan_sip.c: Header 14 [ 0]: [Jul 6 16:52:37] DEBUG[5870] chan_sip.c: Body 0 [ 3]: v=0 [Jul 6 16:52:37] DEBUG[5870] chan_sip.c: Body 1 [ 44]: o=root 909065915 909065915 IN IP4 10.10.6.46 [Jul 6 16:52:37] DEBUG[5870] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 1.6.2.9 [Jul 6 16:52:37] DEBUG[5870] chan_sip.c: Body 3 [ 19]: c=IN IP4 10.10.6.46 [Jul 6 16:52:37] DEBUG[5870] chan_sip.c: Body 4 [ 5]: t=0 0 [Jul 6 16:52:37] DEBUG[5870] chan_sip.c: Body 5 [ 31]: m=audio 15326 RTP/AVP 0 8 3 101 [Jul 6 16:52:37] DEBUG[5870] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Jul 6 16:52:37] DEBUG[5870] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Jul 6 16:52:37] DEBUG[5870] chan_sip.c: Body 8 [ 19]: a=rtpmap:3 GSM/8000 [Jul 6 16:52:37] DEBUG[5870] chan_sip.c: Body 9 [ 33]: a=rtpmap:101 telephone-event/8000 [Jul 6 16:52:37] DEBUG[5870] chan_sip.c: Body 10 [ 15]: a=fmtp:101 0-16 [Jul 6 16:52:37] DEBUG[5870] chan_sip.c: Body 11 [ 10]: a=ptime:20 [Jul 6 16:52:37] DEBUG[5870] chan_sip.c: Body 12 [ 10]: a=sendrecv [Jul 6 16:52:37] VERBOSE[5870] chan_sip.c: Reliably Transmitting (no NAT) to 10.10.10.16:5060: INVITE sip:10.10.10.16:5060 SIP/2.0 Via: SIP/2.0/UDP 10.10.6.46:5060;branch=z9hG4bK298d4a7a;rport Max-Forwards: 70 From: "asterisk" ;tag=as317a6b87 To: Contact: Call-ID: 04f8bd1d4d261c8d4fbb3a0507c9c5ab@10.10.6.46 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.2.9 Date: Tue, 06 Jul 2010 23:52:37 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 277 v=0 o=root 909065915 909065915 IN IP4 10.10.6.46 s=Asterisk PBX 1.6.2.9 c=IN IP4 10.10.6.46 t=0 0 m=audio 15326 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Jul 6 16:52:37] DEBUG[5870] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #18466 [Jul 6 16:52:37] DEBUG[5870] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 10.10.10.16:5060 [Jul 6 16:52:37] VERBOSE[29058] chan_sip.c: <--- SIP read from UDP:10.10.10.16:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.6.46:5060;branch=z9hG4bK298d4a7a;rport From: "asterisk" ;tag=as317a6b87 To: Call-ID: 04f8bd1d4d261c8d4fbb3a0507c9c5ab@10.10.6.46 CSeq: 102 INVITE Supported: timer User-Agent: ShoreTel_FSS_SIP_UATK ShoreTel 7 Contact: DefaultProfile Content-Length: 0 <-------------> [Jul 6 16:52:37] DEBUG[29058] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Jul 6 16:52:37] DEBUG[29058] chan_sip.c: Header 1 [ 61]: Via: SIP/2.0/UDP 10.10.6.46:5060;branch=z9hG4bK298d4a7a;rport [Jul 6 16:52:37] DEBUG[29058] chan_sip.c: Header 2 [ 57]: From: "asterisk" ;tag=as317a6b87 [Jul 6 16:52:37] DEBUG[29058] chan_sip.c: Header 3 [ 26]: To: [Jul 6 16:52:37] DEBUG[29058] chan_sip.c: Header 4 [ 52]: Call-ID: 04f8bd1d4d261c8d4fbb3a0507c9c5ab@10.10.6.46 [Jul 6 16:52:37] DEBUG[29058] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Jul 6 16:52:37] DEBUG[29058] chan_sip.c: Header 6 [ 16]: Supported: timer [Jul 6 16:52:37] DEBUG[29058] chan_sip.c: Header 7 [ 44]: User-Agent: ShoreTel_FSS_SIP_UATK ShoreTel 7 [Jul 6 16:52:37] DEBUG[29058] chan_sip.c: Header 8 [ 61]: Contact: DefaultProfile [Jul 6 16:52:37] DEBUG[29058] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Jul 6 16:52:37] DEBUG[29058] chan_sip.c: Header 10 [ 0]: [Jul 6 16:52:37] VERBOSE[29058] chan_sip.c: --- (10 headers 0 lines) --- [Jul 6 16:52:37] DEBUG[29058] chan_sip.c: *** SIP TIMER: Cancelling retransmission #18466 - INVITE (got response) [Jul 6 16:52:37] DEBUG[29058] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '04f8bd1d4d261c8d4fbb3a0507c9c5ab@10.10.6.46' Request 102: Found [Jul 6 16:52:37] DEBUG[29058] chan_sip.c: SIP response 100 to standard invite [Jul 6 16:52:37] VERBOSE[29058] chan_sip.c: <--- SIP read from UDP:10.10.10.16:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.10.6.46:5060;branch=z9hG4bK298d4a7a;rport From: "asterisk" ;tag=as317a6b87 To: ;tag=hssUA_1201106736-113939383 Call-ID: 04f8bd1d4d261c8d4fbb3a0507c9c5ab@10.10.6.46 CSeq: 102 INVITE Supported: timer User-Agent: ShoreTel_FSS_SIP_UATK ShoreTel 7 Contact: DefaultProfile Content-Length: 0 <-------------> [Jul 6 16:52:37] DEBUG[29058] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Jul 6 16:52:37] DEBUG[29058] chan_sip.c: Header 1 [ 61]: Via: SIP/2.0/UDP 10.10.6.46:5060;branch=z9hG4bK298d4a7a;rport [Jul 6 16:52:37] DEBUG[29058] chan_sip.c: Header 2 [ 57]: From: "asterisk" ;tag=as317a6b87 [Jul 6 16:52:37] DEBUG[29058] chan_sip.c: Header 3 [ 57]: To: ;tag=hssUA_1201106736-113939383 [Jul 6 16:52:37] DEBUG[29058] chan_sip.c: Header 4 [ 52]: Call-ID: 04f8bd1d4d261c8d4fbb3a0507c9c5ab@10.10.6.46 [Jul 6 16:52:37] DEBUG[29058] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Jul 6 16:52:37] DEBUG[29058] chan_sip.c: Header 6 [ 16]: Supported: timer [Jul 6 16:52:37] DEBUG[29058] chan_sip.c: Header 7 [ 44]: User-Agent: ShoreTel_FSS_SIP_UATK ShoreTel 7 [Jul 6 16:52:37] DEBUG[29058] chan_sip.c: Header 8 [ 61]: Contact: DefaultProfile [Jul 6 16:52:37] DEBUG[29058] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Jul 6 16:52:37] DEBUG[29058] chan_sip.c: Header 10 [ 0]: [Jul 6 16:52:37] VERBOSE[29058] chan_sip.c: --- (10 headers 0 lines) --- [Jul 6 16:52:37] DEBUG[29058] chan_sip.c: Acked pending invite 102 [Jul 6 16:52:37] DEBUG[29058] chan_sip.c: Stopping retransmission on '04f8bd1d4d261c8d4fbb3a0507c9c5ab@10.10.6.46' of Request 102: Match Found [Jul 6 16:52:37] DEBUG[29058] chan_sip.c: SIP response 401 to standard invite [Jul 6 16:52:37] VERBOSE[29058] chan_sip.c: Transmitting (no NAT) to 10.10.10.16:5060: ACK sip:10.10.10.16:5060 SIP/2.0 Via: SIP/2.0/UDP 10.10.6.46:5060;branch=z9hG4bK298d4a7a;rport Max-Forwards: 70 From: "asterisk" ;tag=as317a6b87 To: ;tag=hssUA_1201106736-113939383 Contact: Call-ID: 04f8bd1d4d261c8d4fbb3a0507c9c5ab@10.10.6.46 CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.2.9 Content-Length: 0 --- [Jul 6 16:52:37] DEBUG[29058] chan_sip.c: Trying to put 'ACK sip:10.' onto UDP socket destined for 10.10.10.16:5060 [Jul 6 16:52:37] DEBUG[29058] chan_sip.c: Auth attempt 1 on INVITE [Jul 6 16:52:37] NOTICE[29058] chan_sip.c: Failed to authenticate on INVITE to '"asterisk" ;tag=as317a6b87' [Jul 6 16:52:37] DEBUG[29058] chan_sip.c: Setting SIP_ALREADYGONE on dialog 04f8bd1d4d261c8d4fbb3a0507c9c5ab@10.10.6.46 [Jul 6 16:52:37] VERBOSE[5870] pbx.c: > Channel SIP/ShoreTel-00000052 was never answered. [Jul 6 16:52:37] DEBUG[5870] channel.c: Hanging up channel 'SIP/ShoreTel-00000052' [Jul 6 16:52:37] DEBUG[5870] chan_sip.c: Hangup call SIP/ShoreTel-00000052, SIP callid 04f8bd1d4d261c8d4fbb3a0507c9c5ab@10.10.6.46 [Jul 6 16:52:37] DEBUG[5870] chan_sip.c: Hanging up channel in state Down (not UP) [Jul 6 16:52:37] DEBUG[29044] devicestate.c: No provider found, checking channel drivers for SIP - ShoreTel [Jul 6 16:52:37] DEBUG[29044] chan_sip.c: Checking device state for peer ShoreTel [Jul 6 16:52:37] DEBUG[29044] devicestate.c: Changing state for SIP/ShoreTel - state 1 (Not in use) [Jul 6 16:52:37] DEBUG[29044] devicestate.c: device 'SIP/ShoreTel' state '1' [Jul 6 16:52:37] DEBUG[29052] app_queue.c: Device 'SIP/ShoreTel' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jul 6 16:52:38] DEBUG[29058] chan_sip.c: Destroying SIP dialog 04f8bd1d4d261c8d4fbb3a0507c9c5ab@10.10.6.46 [Jul 6 16:52:38] VERBOSE[29058] chan_sip.c: Really destroying SIP dialog '04f8bd1d4d261c8d4fbb3a0507c9c5ab@10.10.6.46' Method: INVITE