[Nov 4 20:17:59] VERBOSE[3578] config.c: == Parsing '/etc/asterisk/logger.conf': [Nov 4 20:17:59] DEBUG[3578] config.c: Parsing /etc/asterisk/logger.conf [Nov 4 20:17:59] VERBOSE[3578] config.c: == Found [Nov 4 20:17:59] VERBOSE[3578] logger.c: Asterisk Queue Logger restarted [Nov 4 20:18:04] DEBUG[3503] chan_sip.c: = Looking for Call ID: 19184f42-1f5ef404-96d8ea23@192.168.10.163 (Checking From) --From tag ACBF54C6-C5EB6CE5 --To-tag [Nov 4 20:18:04] DEBUG[3503] acl.c: For destination '192.168.10.163', our source address is '192.168.10.181'. [Nov 4 20:18:04] DEBUG[3503] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.10.181:5060 [Nov 4 20:18:04] DEBUG[3503] chan_sip.c: Allocating new SIP dialog for 19184f42-1f5ef404-96d8ea23@192.168.10.163 - REGISTER (No RTP) [Nov 4 20:18:04] DEBUG[3503] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Nov 4 20:18:04] DEBUG[3503] netsock2.c: Splitting '192.168.10.163' gives... [Nov 4 20:18:04] DEBUG[3503] netsock2.c: ...host '192.168.10.163' and port '(null)'. [Nov 4 20:18:04] DEBUG[3503] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 192.168.10.163:5060 [Nov 4 20:18:04] DEBUG[3503] chan_sip.c: = Looking for Call ID: 19184f42-1f5ef404-96d8ea23@192.168.10.163 (Checking From) --From tag ACBF54C6-C5EB6CE5 --To-tag [Nov 4 20:18:04] DEBUG[3503] netsock2.c: Splitting '192.168.10.181' gives... [Nov 4 20:18:04] DEBUG[3503] netsock2.c: ...host '192.168.10.181' and port '(null)'. [Nov 4 20:18:04] DEBUG[3503] netsock2.c: Splitting '192.168.10.181' gives... [Nov 4 20:18:04] DEBUG[3503] netsock2.c: ...host '192.168.10.181' and port '(null)'. [Nov 4 20:18:04] DEBUG[3503] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Nov 4 20:18:04] DEBUG[3503] netsock2.c: Splitting '192.168.10.163' gives... [Nov 4 20:18:04] DEBUG[3503] netsock2.c: ...host '192.168.10.163' and port '(null)'. [Nov 4 20:18:04] DEBUG[3503] chan_sip.c: Store REGISTER's Contact header for call routing. [Nov 4 20:18:04] DEBUG[3503] netsock2.c: Splitting '192.168.10.163' gives... [Nov 4 20:18:04] DEBUG[3503] netsock2.c: ...host '192.168.10.163' and port '(null)'. [Nov 4 20:18:04] DEBUG[3508] manager.c: Examining event: Event: PeerStatus Privilege: system,all ChannelType: SIP Peer: SIP/623 PeerStatus: Registered Address: 192.168.10.163:5060 [Nov 4 20:18:04] DEBUG[3503] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.10.163:5060 [Nov 4 20:18:04] DEBUG[3495] devicestate.c: No provider found, checking channel drivers for SIP - 623 [Nov 4 20:18:04] DEBUG[3495] chan_sip.c: Checking device state for peer 623 [Nov 4 20:18:04] DEBUG[3495] devicestate.c: Changing state for SIP/623 - state 1 (Not in use) [Nov 4 20:18:04] DEBUG[3495] devicestate.c: device 'SIP/623' state '1' [Nov 4 20:18:04] DEBUG[3533] app_queue.c: Device 'SIP/623' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Nov 4 20:18:08] DEBUG[3503] chan_sip.c: = Looking for Call ID: 31383749-3d40c097@192.168.10.154 (Checking From) --From tag 4bebcde1e3145adfo0 --To-tag [Nov 4 20:18:08] DEBUG[3503] chan_sip.c: **** Received NOTIFY (4) - Command in SIP NOTIFY [Nov 4 20:18:08] DEBUG[3503] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.10.154:5061 [Nov 4 20:18:14] VERBOSE[3503] chan_sip.c: <--- SIP read from UDP:192.168.10.163:5060 ---> <-------------> [Nov 4 20:18:14] DEBUG[3503] chan_sip.c: Header 0 [ 0]: [Nov 4 20:18:19] VERBOSE[3503] chan_sip.c: <--- SIP read from UDP:192.168.10.163:5060 ---> <-------------> [Nov 4 20:18:19] DEBUG[3503] chan_sip.c: Header 0 [ 0]: [Nov 4 20:18:22] VERBOSE[3503] chan_sip.c: <--- SIP read from UDP:192.168.10.163:5060 ---> INVITE sip:500@192.168.10.181;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.10.163;branch=z9hG4bK6819213cA149E21D From: "623" ;tag=FEC92782-26913BA1 To: CSeq: 1 INVITE Call-ID: 1cfa11fe-cfdcbec0-546ab0df@192.168.10.163 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.2.2.0084 Supported: 100rel,replaces Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 254 v=0 o=- 1288895620 1288895620 IN IP4 192.168.10.163 s=Polycom IP Phone c=IN IP4 192.168.10.163 t=0 0 m=audio 10002 RTP/AVP 0 8 18 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 <-------------> [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: Header 0 [ 48]: INVITE sip:500@192.168.10.181;user=phone SIP/2.0 [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: Header 1 [ 62]: Via: SIP/2.0/UDP 192.168.10.163;branch=z9hG4bK6819213cA149E21D [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: Header 2 [ 58]: From: "623" ;tag=FEC92782-26913BA1 [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: Header 3 [ 39]: To: [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: Header 4 [ 14]: CSeq: 1 INVITE [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: Header 5 [ 50]: Call-ID: 1cfa11fe-cfdcbec0-546ab0df@192.168.10.163 [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: Header 6 [ 33]: Contact: [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: Header 7 [ 96]: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: Header 8 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.2.2.0084 [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: Header 9 [ 26]: Supported: 100rel,replaces [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: Header 10 [ 34]: Allow-Events: talk,hold,conference [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: Header 11 [ 16]: Max-Forwards: 70 [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: Header 13 [ 19]: Content-Length: 254 [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: Header 14 [ 0]: [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: Body 0 [ 3]: v=0 [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: Body 1 [ 47]: o=- 1288895620 1288895620 IN IP4 192.168.10.163 [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: Body 2 [ 18]: s=Polycom IP Phone [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.10.163 [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: Body 4 [ 5]: t=0 0 [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: Body 5 [ 32]: m=audio 10002 RTP/AVP 0 8 18 101 [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: Body 6 [ 10]: a=sendrecv [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: Body 8 [ 20]: a=rtpmap:8 PCMA/8000 [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: Body 9 [ 21]: a=rtpmap:18 G729/8000 [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: Body 10 [ 33]: a=rtpmap:101 telephone-event/8000 [Nov 4 20:18:22] VERBOSE[3503] chan_sip.c: --- (14 headers 11 lines) --- [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: = Looking for Call ID: 1cfa11fe-cfdcbec0-546ab0df@192.168.10.163 (Checking From) --From tag FEC92782-26913BA1 --To-tag [Nov 4 20:18:22] DEBUG[3503] acl.c: For destination '192.168.10.163', our source address is '192.168.10.181'. [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.10.181:5060 [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: Allocating new SIP dialog for 1cfa11fe-cfdcbec0-546ab0df@192.168.10.163 - INVITE (No RTP) [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Nov 4 20:18:22] DEBUG[3503] sip/reqresp_parser.c: Begin: parsing SIP "Supported: 100rel,replaces" [Nov 4 20:18:22] DEBUG[3503] sip/reqresp_parser.c: Found SIP option: -100rel- [Nov 4 20:18:22] DEBUG[3503] sip/reqresp_parser.c: Matched SIP option: 100rel [Nov 4 20:18:22] DEBUG[3503] sip/reqresp_parser.c: Found SIP option: -replaces- [Nov 4 20:18:22] DEBUG[3503] sip/reqresp_parser.c: Matched SIP option: replaces [Nov 4 20:18:22] DEBUG[3503] netsock2.c: Splitting '192.168.10.163' gives... [Nov 4 20:18:22] DEBUG[3503] netsock2.c: ...host '192.168.10.163' and port '(null)'. [Nov 4 20:18:22] VERBOSE[3503] chan_sip.c: Sending to 192.168.10.163:5060 (no NAT) [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: Initializing initreq for method INVITE - callid 1cfa11fe-cfdcbec0-546ab0df@192.168.10.163 [Nov 4 20:18:22] VERBOSE[3503] chan_sip.c: Using INVITE request as basis request - 1cfa11fe-cfdcbec0-546ab0df@192.168.10.163 [Nov 4 20:18:22] VERBOSE[3503] chan_sip.c: Found peer '623' for '623' from 192.168.10.163:5060 [Nov 4 20:18:22] VERBOSE[3503] chan_sip.c: <--- Reliably Transmitting (no NAT) to 192.168.10.163:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.10.163;branch=z9hG4bK6819213cA149E21D;received=192.168.10.163 From: "623" ;tag=FEC92782-26913BA1 To: ;tag=as0b8b0dc1 Call-ID: 1cfa11fe-cfdcbec0-546ab0df@192.168.10.163 CSeq: 1 INVITE Server: Asterisk PBX 1.8.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6a08ddaa" Content-Length: 0 <------------> [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #300 [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 192.168.10.163:5060 [Nov 4 20:18:22] VERBOSE[3503] chan_sip.c: Scheduling destruction of SIP dialog '1cfa11fe-cfdcbec0-546ab0df@192.168.10.163' in 32000 ms (Method: INVITE) [Nov 4 20:18:22] VERBOSE[3503] chan_sip.c: <--- SIP read from UDP:192.168.10.163:5060 ---> ACK sip:500@192.168.10.181 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.163;branch=z9hG4bK6819213cA149E21D From: "623" ;tag=FEC92782-26913BA1 To: ;tag=as0b8b0dc1 CSeq: 1 ACK Call-ID: 1cfa11fe-cfdcbec0-546ab0df@192.168.10.163 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.2.2.0084 Max-Forwards: 70 Content-Length: 0 <-------------> [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: Header 0 [ 34]: ACK sip:500@192.168.10.181 SIP/2.0 [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: Header 1 [ 62]: Via: SIP/2.0/UDP 192.168.10.163;branch=z9hG4bK6819213cA149E21D [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: Header 2 [ 58]: From: "623" ;tag=FEC92782-26913BA1 [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: Header 3 [ 54]: To: ;tag=as0b8b0dc1 [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: Header 4 [ 11]: CSeq: 1 ACK [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: Header 5 [ 50]: Call-ID: 1cfa11fe-cfdcbec0-546ab0df@192.168.10.163 [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: Header 6 [ 33]: Contact: [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: Header 7 [ 96]: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: Header 8 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.2.2.0084 [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: Header 9 [ 16]: Max-Forwards: 70 [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Nov 4 20:18:22] VERBOSE[3503] chan_sip.c: --- (11 headers 0 lines) --- [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: = Looking for Call ID: 1cfa11fe-cfdcbec0-546ab0df@192.168.10.163 (Checking From) --From tag FEC92782-26913BA1 --To-tag as0b8b0dc1 [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #300 [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: Stopping retransmission on '1cfa11fe-cfdcbec0-546ab0df@192.168.10.163' of Response 1: Match Found [Nov 4 20:18:22] VERBOSE[3503] chan_sip.c: <--- SIP read from UDP:192.168.10.163:5060 ---> INVITE sip:500@192.168.10.181;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.10.163;branch=z9hG4bKdc5826317C74C44 From: "623" ;tag=FEC92782-26913BA1 To: CSeq: 2 INVITE Call-ID: 1cfa11fe-cfdcbec0-546ab0df@192.168.10.163 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.2.2.0084 Supported: 100rel,replaces Allow-Events: talk,hold,conference Authorization: Digest username="623", realm="asterisk", nonce="6a08ddaa", uri="sip:500@192.168.10.181;user=phone", response="aefbd9bfcf61a3201008b6827be3f8bb", algorithm=MD5 Max-Forwards: 70 Content-Type: application/sdp Content-Length: 254 v=0 o=- 1288895620 1288895620 IN IP4 192.168.10.163 s=Polycom IP Phone c=IN IP4 192.168.10.163 t=0 0 m=audio 10002 RTP/AVP 0 8 18 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 <-------------> [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: Header 0 [ 48]: INVITE sip:500@192.168.10.181;user=phone SIP/2.0 [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: Header 1 [ 61]: Via: SIP/2.0/UDP 192.168.10.163;branch=z9hG4bKdc5826317C74C44 [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: Header 2 [ 58]: From: "623" ;tag=FEC92782-26913BA1 [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: Header 3 [ 39]: To: [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: Header 4 [ 14]: CSeq: 2 INVITE [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: Header 5 [ 50]: Call-ID: 1cfa11fe-cfdcbec0-546ab0df@192.168.10.163 [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: Header 6 [ 33]: Contact: [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: Header 7 [ 96]: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: Header 8 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.2.2.0084 [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: Header 9 [ 26]: Supported: 100rel,replaces [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: Header 10 [ 34]: Allow-Events: talk,hold,conference [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: Header 11 [173]: Authorization: Digest username="623", realm="asterisk", nonce="6a08ddaa", uri="sip:500@192.168.10.181;user=phone", response="aefbd9bfcf61a3201008b6827be3f8bb", algorithm=MD5 [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: Header 12 [ 16]: Max-Forwards: 70 [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: Header 14 [ 19]: Content-Length: 254 [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: Header 15 [ 0]: [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: Body 0 [ 3]: v=0 [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: Body 1 [ 47]: o=- 1288895620 1288895620 IN IP4 192.168.10.163 [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: Body 2 [ 18]: s=Polycom IP Phone [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.10.163 [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: Body 4 [ 5]: t=0 0 [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: Body 5 [ 32]: m=audio 10002 RTP/AVP 0 8 18 101 [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: Body 6 [ 10]: a=sendrecv [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: Body 8 [ 20]: a=rtpmap:8 PCMA/8000 [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: Body 9 [ 21]: a=rtpmap:18 G729/8000 [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: Body 10 [ 33]: a=rtpmap:101 telephone-event/8000 [Nov 4 20:18:22] VERBOSE[3503] chan_sip.c: --- (15 headers 11 lines) --- [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: = Looking for Call ID: 1cfa11fe-cfdcbec0-546ab0df@192.168.10.163 (Checking From) --From tag FEC92782-26913BA1 --To-tag [Nov 4 20:18:22] DEBUG[3503] netsock2.c: Splitting '192.168.10.181' gives... [Nov 4 20:18:22] DEBUG[3503] netsock2.c: ...host '192.168.10.181' and port '(null)'. [Nov 4 20:18:22] DEBUG[3503] netsock2.c: Splitting '192.168.10.181' gives... [Nov 4 20:18:22] DEBUG[3503] netsock2.c: ...host '192.168.10.181' and port '(null)'. [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Nov 4 20:18:22] DEBUG[3503] netsock2.c: Splitting '192.168.10.163' gives... [Nov 4 20:18:22] DEBUG[3503] netsock2.c: ...host '192.168.10.163' and port '(null)'. [Nov 4 20:18:22] VERBOSE[3503] chan_sip.c: Sending to 192.168.10.163:5060 (no NAT) [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: Initializing initreq for method INVITE - callid 1cfa11fe-cfdcbec0-546ab0df@192.168.10.163 [Nov 4 20:18:22] VERBOSE[3503] chan_sip.c: Using INVITE request as basis request - 1cfa11fe-cfdcbec0-546ab0df@192.168.10.163 [Nov 4 20:18:22] VERBOSE[3503] chan_sip.c: Found peer '623' for '623' from 192.168.10.163:5060 [Nov 4 20:18:22] DEBUG[3503] rtp_engine.c: Using engine 'asterisk' for RTP instance '0xd320888' [Nov 4 20:18:22] DEBUG[3503] res_rtp_asterisk.c: Allocated port 11684 for RTP instance '0xd320888' [Nov 4 20:18:22] DEBUG[3503] rtp_engine.c: RTP instance '0xd320888' is setup and ready to go [Nov 4 20:18:22] DEBUG[3503] res_rtp_asterisk.c: Setup RTCP on RTP instance '0xd320888' [Nov 4 20:18:22] VERBOSE[3503] netsock2.c: == Using SIP RTP CoS mark 5 [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: Setting NAT on RTP to Off [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: Processing session-level SDP o=- 1288895620 1288895620 IN IP4 192.168.10.163... UNSUPPORTED. [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: Processing session-level SDP s=Polycom IP Phone... UNSUPPORTED. [Nov 4 20:18:22] DEBUG[3503] netsock2.c: Splitting '192.168.10.163' gives... [Nov 4 20:18:22] DEBUG[3503] netsock2.c: ...host '192.168.10.163' and port '(null)'. [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.10.163... OK. [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Nov 4 20:18:22] VERBOSE[3503] chan_sip.c: Found RTP audio format 0 [Nov 4 20:18:22] DEBUG[3503] rtp_engine.c: Setting payload 0 based on m type on 0xb75dd924 [Nov 4 20:18:22] VERBOSE[3503] chan_sip.c: Found RTP audio format 8 [Nov 4 20:18:22] DEBUG[3503] rtp_engine.c: Setting payload 8 based on m type on 0xb75dd924 [Nov 4 20:18:22] VERBOSE[3503] chan_sip.c: Found RTP audio format 18 [Nov 4 20:18:22] DEBUG[3503] rtp_engine.c: Setting payload 18 based on m type on 0xb75dd924 [Nov 4 20:18:22] VERBOSE[3503] chan_sip.c: Found RTP audio format 101 [Nov 4 20:18:22] DEBUG[3503] rtp_engine.c: Setting payload 101 based on m type on 0xb75dd924 [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Nov 4 20:18:22] VERBOSE[3503] chan_sip.c: Found audio description format PCMU for ID 0 [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Nov 4 20:18:22] VERBOSE[3503] chan_sip.c: Found audio description format PCMA for ID 8 [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Nov 4 20:18:22] VERBOSE[3503] chan_sip.c: Found audio description format G729 for ID 18 [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:18 G729/8000... OK. [Nov 4 20:18:22] VERBOSE[3503] chan_sip.c: Found audio description format telephone-event for ID 101 [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Nov 4 20:18:22] DEBUG[3503] rtp_engine.c: Incorporating payload 0 on 0xb75dd924 [Nov 4 20:18:22] DEBUG[3503] rtp_engine.c: Incorporating payload 8 on 0xb75dd924 [Nov 4 20:18:22] DEBUG[3503] rtp_engine.c: Incorporating payload 18 on 0xb75dd924 [Nov 4 20:18:22] DEBUG[3503] rtp_engine.c: Incorporating payload 101 on 0xb75dd924 [Nov 4 20:18:22] VERBOSE[3503] chan_sip.c: Capabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) [Nov 4 20:18:22] VERBOSE[3503] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Nov 4 20:18:22] DEBUG[3503] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xd320888' [Nov 4 20:18:22] VERBOSE[3503] chan_sip.c: Peer audio RTP is at port 192.168.10.163:10002 [Nov 4 20:18:22] DEBUG[3503] rtp_engine.c: Copying payload 0 from 0xb75dd924 to 0xd320a34 [Nov 4 20:18:22] DEBUG[3503] rtp_engine.c: Copying payload 8 from 0xb75dd924 to 0xd320a34 [Nov 4 20:18:22] DEBUG[3503] rtp_engine.c: Copying payload 18 from 0xb75dd924 to 0xd320a34 [Nov 4 20:18:22] DEBUG[3503] rtp_engine.c: Copying payload 101 from 0xb75dd924 to 0xd320a34 [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: Checking SIP call limits for device 623 [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: Updating call counter for incoming call [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: Call from peer '623' is 1 out of 2 [Nov 4 20:18:22] DEBUG[3495] devicestate.c: No provider found, checking channel drivers for SIP - 623 [Nov 4 20:18:22] DEBUG[3495] chan_sip.c: Checking device state for peer 623 [Nov 4 20:18:22] DEBUG[3495] devicestate.c: Changing state for SIP/623 - state 2 (In use) [Nov 4 20:18:22] DEBUG[3495] devicestate.c: device 'SIP/623' state '2' [Nov 4 20:18:22] DEBUG[3533] app_queue.c: Device 'SIP/623' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Nov 4 20:18:22] VERBOSE[3503] chan_sip.c: Looking for 500 in internal (domain 192.168.10.181) [Nov 4 20:18:22] DEBUG[3508] manager.c: Examining event: Event: Newchannel Privilege: call,all Channel: SIP/623-00000010 ChannelState: 0 ChannelStateDesc: Down CallerIDNum: 623 CallerIDName: polycom AccountCode: Exten: 500 Context: internal Uniqueid: 1288898302.16 [Nov 4 20:18:22] DEBUG[3503] frame.c: Could not find preferred codec - Going for the best codec [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: *** Our native formats are 0x4 (ulaw) [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: *** Joint capabilities are 0xc (ulaw|alaw) [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: *** Our capabilities are 0x80000008000e (gsm|ulaw|alaw|h263|testlaw) [Nov 4 20:18:22] DEBUG[3503] frame.c: Could not find preferred codec - Going for the best codec [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: This channel will not be able to handle video. [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: build_route: Contact hop: [Nov 4 20:18:22] VERBOSE[3503] chan_sip.c: list_route: hop: [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: SIP/623-00000010: New call is still down.... Trying... [Nov 4 20:18:22] VERBOSE[3503] chan_sip.c: <--- Transmitting (no NAT) to 192.168.10.163:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.10.163;branch=z9hG4bKdc5826317C74C44;received=192.168.10.163 From: "623" ;tag=FEC92782-26913BA1 To: Call-ID: 1cfa11fe-cfdcbec0-546ab0df@192.168.10.163 CSeq: 2 INVITE Server: Asterisk PBX 1.8.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 192.168.10.163:5060 [Nov 4 20:18:22] DEBUG[3495] devicestate.c: No provider found, checking channel drivers for SIP - 623 [Nov 4 20:18:22] DEBUG[3495] chan_sip.c: Checking device state for peer 623 [Nov 4 20:18:22] DEBUG[3495] devicestate.c: Changing state for SIP/623 - state 2 (In use) [Nov 4 20:18:22] DEBUG[3495] devicestate.c: device 'SIP/623' state '2' [Nov 4 20:18:22] DEBUG[3533] app_queue.c: Device 'SIP/623' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Nov 4 20:18:22] DEBUG[3508] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/623-00000010 ChannelState: 4 ChannelStateDesc: Ring CallerIDNum: 623 CallerIDName: polycom Uniqueid: 1288898302.16 [Nov 4 20:18:22] DEBUG[3585] pbx.c: Launching 'Answer' [Nov 4 20:18:22] VERBOSE[3585] pbx.c: -- Executing [500@internal:1] Answer("SIP/623-00000010", "") in new stack [Nov 4 20:18:22] DEBUG[3495] devicestate.c: No provider found, checking channel drivers for SIP - 623 [Nov 4 20:18:22] DEBUG[3495] chan_sip.c: Checking device state for peer 623 [Nov 4 20:18:22] DEBUG[3495] devicestate.c: Changing state for SIP/623 - state 2 (In use) [Nov 4 20:18:22] DEBUG[3495] devicestate.c: device 'SIP/623' state '2' [Nov 4 20:18:22] DEBUG[3533] app_queue.c: Device 'SIP/623' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Nov 4 20:18:22] DEBUG[3508] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/623-00000010 ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 623 CallerIDName: polycom Uniqueid: 1288898302.16 [Nov 4 20:18:22] DEBUG[3585] chan_sip.c: SIP answering channel: SIP/623-00000010 [Nov 4 20:18:22] DEBUG[3585] res_rtp_asterisk.c: Setting the marker bit due to a source update [Nov 4 20:18:22] DEBUG[3585] chan_sip.c: Setting framing from config on incoming call [Nov 4 20:18:22] DEBUG[3585] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: True Text flag: True [Nov 4 20:18:22] DEBUG[3585] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Nov 4 20:18:22] VERBOSE[3585] chan_sip.c: Audio is at 5060 [Nov 4 20:18:22] VERBOSE[3585] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Nov 4 20:18:22] VERBOSE[3585] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Nov 4 20:18:22] VERBOSE[3585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Nov 4 20:18:22] DEBUG[3585] chan_sip.c: -- Done with adding codecs to SDP [Nov 4 20:18:22] DEBUG[3585] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Nov 4 20:18:22] VERBOSE[3585] chan_sip.c: <--- Reliably Transmitting (no NAT) to 192.168.10.163:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.163;branch=z9hG4bKdc5826317C74C44;received=192.168.10.163 From: "623" ;tag=FEC92782-26913BA1 To: ;tag=as7394ab0e Call-ID: 1cfa11fe-cfdcbec0-546ab0df@192.168.10.163 CSeq: 2 INVITE Server: Asterisk PBX 1.8.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 262 v=0 o=root 1286878123 1286878123 IN IP4 192.168.10.181 s=Asterisk PBX 1.8.0 c=IN IP4 192.168.10.181 t=0 0 m=audio 11684 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <------------> [Nov 4 20:18:22] DEBUG[3585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #303 [Nov 4 20:18:22] DEBUG[3585] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.10.163:5060 [Nov 4 20:18:22] VERBOSE[3503] chan_sip.c: <--- SIP read from UDP:192.168.10.163:5060 ---> ACK sip:500@192.168.10.181:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.163;branch=z9hG4bK7018c9c8EE04BEA9 From: "623" ;tag=FEC92782-26913BA1 To: ;tag=as7394ab0e CSeq: 2 ACK Call-ID: 1cfa11fe-cfdcbec0-546ab0df@192.168.10.163 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.2.2.0084 Authorization: Digest username="623", realm="asterisk", nonce="6a08ddaa", uri="sip:500@192.168.10.181;user=phone", response="aefbd9bfcf61a3201008b6827be3f8bb", algorithm=MD5 Max-Forwards: 70 Content-Length: 0 <-------------> [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: Header 0 [ 39]: ACK sip:500@192.168.10.181:5060 SIP/2.0 [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: Header 1 [ 62]: Via: SIP/2.0/UDP 192.168.10.163;branch=z9hG4bK7018c9c8EE04BEA9 [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: Header 2 [ 58]: From: "623" ;tag=FEC92782-26913BA1 [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: Header 3 [ 54]: To: ;tag=as7394ab0e [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: Header 4 [ 11]: CSeq: 2 ACK [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: Header 5 [ 50]: Call-ID: 1cfa11fe-cfdcbec0-546ab0df@192.168.10.163 [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: Header 6 [ 33]: Contact: [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: Header 7 [ 96]: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: Header 8 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.2.2.0084 [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: Header 9 [173]: Authorization: Digest username="623", realm="asterisk", nonce="6a08ddaa", uri="sip:500@192.168.10.181;user=phone", response="aefbd9bfcf61a3201008b6827be3f8bb", algorithm=MD5 [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: Header 10 [ 16]: Max-Forwards: 70 [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Nov 4 20:18:22] VERBOSE[3503] chan_sip.c: --- (12 headers 0 lines) --- [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: = Looking for Call ID: 1cfa11fe-cfdcbec0-546ab0df@192.168.10.163 (Checking From) --From tag FEC92782-26913BA1 --To-tag as7394ab0e [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #303 [Nov 4 20:18:22] DEBUG[3503] chan_sip.c: Stopping retransmission on '1cfa11fe-cfdcbec0-546ab0df@192.168.10.163' of Response 2: Match Found [Nov 4 20:18:22] DEBUG[3585] pbx.c: Launching 'Goto' [Nov 4 20:18:22] VERBOSE[3585] pbx.c: -- Executing [500@internal:2] Goto("SIP/623-00000010", "do_support,s,1") in new stack [Nov 4 20:18:22] VERBOSE[3585] pbx.c: -- Goto (do_support,s,1) [Nov 4 20:18:22] DEBUG[3585] pbx.c: Launching 'Wait' [Nov 4 20:18:22] VERBOSE[3585] pbx.c: -- Executing [s@do_support:1] Wait("SIP/623-00000010", "0.5") in new stack [Nov 4 20:18:23] DEBUG[3585] pbx.c: Launching 'Set' [Nov 4 20:18:23] VERBOSE[3585] pbx.c: -- Executing [s@do_support:2] Set("SIP/623-00000010", "CALLERID(name)=MR X") in new stack [Nov 4 20:18:23] DEBUG[3585] pbx.c: Launching 'Set' [Nov 4 20:18:23] VERBOSE[3585] pbx.c: -- Executing [s@do_support:3] Set("SIP/623-00000010", "CALLERID(num)=12345") in new stack [Nov 4 20:18:23] DEBUG[3585] pbx.c: Launching 'Queue' [Nov 4 20:18:23] VERBOSE[3585] pbx.c: -- Executing [s@do_support:4] Queue("SIP/623-00000010", "queue_support,tT,,,1800") in new stack [Nov 4 20:18:23] DEBUG[3585] app_queue.c: NO QUEUE_PRIO variable found. Using default. [Nov 4 20:18:23] DEBUG[3585] app_queue.c: queue: queue_support, options: tT, url: , announce: , expires: 1288900103, priority: 0 [Nov 4 20:18:23] DEBUG[3585] app_queue.c: Queue queue_support has no realtime members defined. No need for update [Nov 4 20:18:23] DEBUG[3508] manager.c: Examining event: Event: Join Privilege: call,all Channel: SIP/623-00000010 CallerIDNum: 12345 CallerIDName: MR X Queue: queue_support Position: 1 Count: 1 Uniqueid: 1288898302.16 [Nov 4 20:18:23] DEBUG[3585] app_queue.c: Queue 'queue_support' Join, Channel 'SIP/623-00000010', Position '1' [Nov 4 20:18:23] DEBUG[3508] manager.c: Examining event: Event: MusicOnHold Privilege: call,all State: Start Channel: SIP/623-00000010 UniqueID: 1288898302.16 Class: default [Nov 4 20:18:23] VERBOSE[3585] res_musiconhold.c: -- Started music on hold, class 'default', on SIP/623-00000010 [Nov 4 20:18:23] DEBUG[3585] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Nov 4 20:18:23] DEBUG[3585] app_queue.c: There is 1 available member. [Nov 4 20:18:23] DEBUG[3585] app_queue.c: It's our turn (SIP/623-00000010). [Nov 4 20:18:23] DEBUG[3585] app_queue.c: SIP/623-00000010 is trying to call a queue member. [Nov 4 20:18:23] DEBUG[3585] app_queue.c: (Parallel) Trying 'SIP/671' with metric 0 [Nov 4 20:18:23] DEBUG[3585] chan_sip.c: Asked to create a SIP channel with formats: 0x4 (ulaw) [Nov 4 20:18:23] DEBUG[3585] chan_sip.c: Allocating new SIP dialog for 6349f63f378696a103cca0e3123b92d2@127.0.1.1:0 - INVITE (No RTP) [Nov 4 20:18:23] DEBUG[3585] rtp_engine.c: Using engine 'asterisk' for RTP instance '0xd2c3798' [Nov 4 20:18:23] DEBUG[3585] res_rtp_asterisk.c: Allocated port 13534 for RTP instance '0xd2c3798' [Nov 4 20:18:23] DEBUG[3585] rtp_engine.c: RTP instance '0xd2c3798' is setup and ready to go [Nov 4 20:18:23] DEBUG[3585] res_rtp_asterisk.c: Setup RTCP on RTP instance '0xd2c3798' [Nov 4 20:18:23] VERBOSE[3585] netsock2.c: == Using SIP RTP CoS mark 5 [Nov 4 20:18:23] DEBUG[3585] chan_sip.c: Setting NAT on RTP to Off [Nov 4 20:18:23] DEBUG[3585] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Nov 4 20:18:23] DEBUG[3585] acl.c: For destination '192.168.10.10', our source address is '192.168.10.181'. [Nov 4 20:18:23] DEBUG[3585] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.10.181:5060 [Nov 4 20:18:23] DEBUG[3508] manager.c: Examining event: Event: Newchannel Privilege: call,all Channel: SIP/671-00000011 ChannelState: 0 ChannelStateDesc: Down CallerIDNum: 671 CallerIDName: 3CX Phone AccountCode: Exten: Context: internal Uniqueid: 1288898303.17 [Nov 4 20:18:23] DEBUG[3585] frame.c: Could not find preferred codec - Going for the best codec [Nov 4 20:18:23] DEBUG[3585] chan_sip.c: *** Our native formats are 0x4 (ulaw) [Nov 4 20:18:23] DEBUG[3585] chan_sip.c: *** Joint capabilities are 0x4 (ulaw) [Nov 4 20:18:23] DEBUG[3585] chan_sip.c: *** Our capabilities are 0x80000008000e (gsm|ulaw|alaw|h263|testlaw) [Nov 4 20:18:23] DEBUG[3585] frame.c: Could not find preferred codec - Going for the best codec [Nov 4 20:18:23] DEBUG[3585] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [Nov 4 20:18:23] DEBUG[3585] chan_sip.c: *** Our preferred formats from the incoming channel are 0x4 (ulaw) [Nov 4 20:18:23] DEBUG[3585] chan_sip.c: This channel will not be able to handle video. [Nov 4 20:18:23] DEBUG[3585] channel.c: Not copying variable SIPCALLID. [Nov 4 20:18:23] DEBUG[3585] channel.c: Not copying variable SIPDOMAIN. [Nov 4 20:18:23] DEBUG[3585] channel.c: Not copying variable SIPURI. [Nov 4 20:18:23] DEBUG[3585] chan_sip.c: Outgoing Call for 671 [Nov 4 20:18:23] DEBUG[3585] chan_sip.c: Updating call counter for outgoing call [Nov 4 20:18:23] DEBUG[3585] chan_sip.c: Call to peer '671' is 1 out of 2 [Nov 4 20:18:23] DEBUG[3495] devicestate.c: No provider found, checking channel drivers for SIP - 671 [Nov 4 20:18:23] DEBUG[3495] chan_sip.c: Checking device state for peer 671 [Nov 4 20:18:23] DEBUG[3495] devicestate.c: Changing state for SIP/671 - state 6 (Ringing) [Nov 4 20:18:23] DEBUG[3495] devicestate.c: device 'SIP/671' state '6' [Nov 4 20:18:23] DEBUG[3533] app_queue.c: Device 'SIP/671' changed to state '6' (Ringing) [Nov 4 20:18:23] DEBUG[3508] manager.c: Examining event: Event: QueueMemberStatus Privilege: agent,all Queue: queue_support Location: SIP/671 MemberName: SIP/671 Membership: static Penalty: 0 CallsTaken: 1 LastCall: 1288898028 Status: 6 Paused: 0 [Nov 4 20:18:23] DEBUG[3585] chan_sip.c: This call needs video offers, but there's no video support enabled! [Nov 4 20:18:23] DEBUG[3585] chan_sip.c: ** Our capability: 0x80000008000e (gsm|ulaw|alaw|h263|testlaw) Video flag: False Text flag: False [Nov 4 20:18:23] DEBUG[3585] chan_sip.c: ** Our prefcodec: 0x4 (ulaw) [Nov 4 20:18:23] VERBOSE[3585] chan_sip.c: Audio is at 5060 [Nov 4 20:18:23] VERBOSE[3585] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Nov 4 20:18:23] VERBOSE[3585] chan_sip.c: Adding codec 0x2 (gsm) to SDP [Nov 4 20:18:23] VERBOSE[3585] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Nov 4 20:18:23] VERBOSE[3585] chan_sip.c: Adding codec 0x800000000000 (testlaw) to SDP [Nov 4 20:18:23] VERBOSE[3585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Nov 4 20:18:23] DEBUG[3585] chan_sip.c: -- Done with adding codecs to SDP [Nov 4 20:18:23] DEBUG[3585] chan_sip.c: Done building SDP. Settling with this capability: 0x80000008000e (gsm|ulaw|alaw|h263|testlaw) [Nov 4 20:18:23] DEBUG[3585] chan_sip.c: Initializing initreq for method INVITE - callid 70fcefbd1612892f47d24c6229e92e83@192.168.10.181:5060 [Nov 4 20:18:23] DEBUG[3585] chan_sip.c: Header 0 [ 69]: INVITE sip:671@192.168.10.10:51944;rinstance=29da946aeae5fdb8 SIP/2.0 [Nov 4 20:18:23] DEBUG[3585] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/UDP 192.168.10.181:5060;branch=z9hG4bK02310c25 [Nov 4 20:18:23] DEBUG[3585] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 4 20:18:23] DEBUG[3585] chan_sip.c: Header 3 [ 54]: From: "MR X" ;tag=as34507cfd [Nov 4 20:18:23] DEBUG[3585] chan_sip.c: Header 4 [ 60]: To: [Nov 4 20:18:23] DEBUG[3585] chan_sip.c: Header 5 [ 40]: Contact: [Nov 4 20:18:23] DEBUG[3585] chan_sip.c: Header 6 [ 61]: Call-ID: 70fcefbd1612892f47d24c6229e92e83@192.168.10.181:5060 [Nov 4 20:18:23] DEBUG[3585] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Nov 4 20:18:23] DEBUG[3585] chan_sip.c: Header 8 [ 30]: User-Agent: Asterisk PBX 1.8.0 [Nov 4 20:18:23] DEBUG[3585] chan_sip.c: Header 9 [ 35]: Date: Thu, 04 Nov 2010 19:18:23 GMT [Nov 4 20:18:23] DEBUG[3585] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 4 20:18:23] DEBUG[3585] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Nov 4 20:18:23] DEBUG[3585] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Nov 4 20:18:23] VERBOSE[3585] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.10.10:51944: INVITE sip:671@192.168.10.10:51944;rinstance=29da946aeae5fdb8 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.181:5060;branch=z9hG4bK02310c25 Max-Forwards: 70 From: "MR X" ;tag=as34507cfd To: Contact: Call-ID: 70fcefbd1612892f47d24c6229e92e83@192.168.10.181:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.0 Date: Thu, 04 Nov 2010 19:18:23 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 285 v=0 o=root 1705840137 1705840137 IN IP4 192.168.10.181 s=Asterisk PBX 1.8.0 c=IN IP4 192.168.10.181 t=0 0 m=audio 13534 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Nov 4 20:18:23] DEBUG[3585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #305 [Nov 4 20:18:23] DEBUG[3585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.10.10:51944 [Nov 4 20:18:23] DEBUG[3508] manager.c: Examining event: Event: QueueMemberStatus Privilege: agent,all Queue: queue_support Location: SIP/671 MemberName: SIP/671 Membership: static Penalty: 0 CallsTaken: 1 LastCall: 1288898028 Status: 6 Paused: 0 [Nov 4 20:18:23] DEBUG[3585] channel.c: Set channel SIP/623-00000010 to write format slin [Nov 4 20:18:23] DEBUG[3585] res_musiconhold.c: SIP/623-00000010 Opened file 0 '/var/lib/asterisk/moh/manolo_camp-morning_coffee' [Nov 4 20:18:23] DEBUG[3585] res_rtp_asterisk.c: Ooh, format changed from unknown to ulaw [Nov 4 20:18:23] DEBUG[3585] res_rtp_asterisk.c: Created smoother: format: ulaw ms: 20 len: 160 [Nov 4 20:18:23] DEBUG[3503] chan_sip.c: SIP TIMER: Rescheduling retransmission #305 (1) INVITE - 5 [Nov 4 20:18:23] DEBUG[3503] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 220 ms (t1 110 ms (Retrans id #305)) [Nov 4 20:18:23] VERBOSE[3503] chan_sip.c: Retransmitting #1 (no NAT) to 192.168.10.10:51944: INVITE sip:671@192.168.10.10:51944;rinstance=29da946aeae5fdb8 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.181:5060;branch=z9hG4bK02310c25 Max-Forwards: 70 From: "MR X" ;tag=as34507cfd To: Contact: Call-ID: 70fcefbd1612892f47d24c6229e92e83@192.168.10.181:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.0 Date: Thu, 04 Nov 2010 19:18:23 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 285 v=0 o=root 1705840137 1705840137 IN IP4 192.168.10.181 s=Asterisk PBX 1.8.0 c=IN IP4 192.168.10.181 t=0 0 m=audio 13534 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Nov 4 20:18:23] DEBUG[3503] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.10.10:51944 [Nov 4 20:18:23] VERBOSE[3503] chan_sip.c: <--- SIP read from UDP:192.168.10.10:51944 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.10.181:5060;branch=z9hG4bK02310c25 Contact: To: ;tag=c2778968 From: "MR X";tag=as34507cfd Call-ID: 70fcefbd1612892f47d24c6229e92e83@192.168.10.181:5060 CSeq: 102 INVITE User-Agent: 3CXPhone 4.0.10858.0 Content-Length: 0 <-------------> [Nov 4 20:18:23] DEBUG[3503] chan_sip.c: Header 0 [ 19]: SIP/2.0 180 Ringing [Nov 4 20:18:23] DEBUG[3503] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/UDP 192.168.10.181:5060;branch=z9hG4bK02310c25 [Nov 4 20:18:23] DEBUG[3503] chan_sip.c: Header 2 [ 65]: Contact: [Nov 4 20:18:23] DEBUG[3503] chan_sip.c: Header 3 [ 73]: To: ;tag=c2778968 [Nov 4 20:18:23] DEBUG[3503] chan_sip.c: Header 4 [ 53]: From: "MR X";tag=as34507cfd [Nov 4 20:18:23] DEBUG[3503] chan_sip.c: Header 5 [ 61]: Call-ID: 70fcefbd1612892f47d24c6229e92e83@192.168.10.181:5060 [Nov 4 20:18:23] DEBUG[3503] chan_sip.c: Header 6 [ 16]: CSeq: 102 INVITE [Nov 4 20:18:23] DEBUG[3503] chan_sip.c: Header 7 [ 32]: User-Agent: 3CXPhone 4.0.10858.0 [Nov 4 20:18:23] DEBUG[3503] chan_sip.c: Header 8 [ 17]: Content-Length: 0 [Nov 4 20:18:23] VERBOSE[3503] chan_sip.c: --- (9 headers 0 lines) --- [Nov 4 20:18:23] DEBUG[3503] chan_sip.c: = Looking for Call ID: 70fcefbd1612892f47d24c6229e92e83@192.168.10.181:5060 (Checking To) --From tag as34507cfd --To-tag c2778968 [Nov 4 20:18:23] DEBUG[3503] chan_sip.c: *** SIP TIMER: Cancelling retransmission #305 - INVITE (got response) [Nov 4 20:18:23] DEBUG[3503] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '70fcefbd1612892f47d24c6229e92e83@192.168.10.181:5060' Request 102: Found [Nov 4 20:18:23] DEBUG[3503] chan_sip.c: SIP response 180 to standard invite [Nov 4 20:18:23] DEBUG[3495] devicestate.c: No provider found, checking channel drivers for SIP - 671 [Nov 4 20:18:23] DEBUG[3495] chan_sip.c: Checking device state for peer 671 [Nov 4 20:18:23] DEBUG[3495] devicestate.c: Changing state for SIP/671 - state 6 (Ringing) [Nov 4 20:18:23] DEBUG[3495] devicestate.c: device 'SIP/671' state '6' [Nov 4 20:18:23] DEBUG[3508] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/671-00000011 ChannelState: 5 ChannelStateDesc: Ringing CallerIDNum: 671 CallerIDName: 3CX Phone Uniqueid: 1288898303.17 [Nov 4 20:18:23] DEBUG[3533] app_queue.c: Device 'SIP/671' changed to state '6' (Ringing) [Nov 4 20:18:23] DEBUG[3508] manager.c: Examining event: Event: QueueMemberStatus Privilege: agent,all Queue: queue_support Location: SIP/671 MemberName: SIP/671 Membership: static Penalty: 0 CallsTaken: 1 LastCall: 1288898028 Status: 6 Paused: 0 [Nov 4 20:18:23] VERBOSE[3585] app_queue.c: -- SIP/671-00000011 is ringing [Nov 4 20:18:23] VERBOSE[3503] chan_sip.c: <--- SIP read from UDP:192.168.10.10:51944 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.10.181:5060;branch=z9hG4bK02310c25 Contact: To: ;tag=c2778968 From: "MR X";tag=as34507cfd Call-ID: 70fcefbd1612892f47d24c6229e92e83@192.168.10.181:5060 CSeq: 102 INVITE User-Agent: 3CXPhone 4.0.10858.0 Content-Length: 0 <-------------> [Nov 4 20:18:23] DEBUG[3503] chan_sip.c: Header 0 [ 19]: SIP/2.0 180 Ringing [Nov 4 20:18:23] DEBUG[3503] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/UDP 192.168.10.181:5060;branch=z9hG4bK02310c25 [Nov 4 20:18:23] DEBUG[3503] chan_sip.c: Header 2 [ 65]: Contact: [Nov 4 20:18:23] DEBUG[3503] chan_sip.c: Header 3 [ 73]: To: ;tag=c2778968 [Nov 4 20:18:23] DEBUG[3503] chan_sip.c: Header 4 [ 53]: From: "MR X";tag=as34507cfd [Nov 4 20:18:23] DEBUG[3503] chan_sip.c: Header 5 [ 61]: Call-ID: 70fcefbd1612892f47d24c6229e92e83@192.168.10.181:5060 [Nov 4 20:18:23] DEBUG[3503] chan_sip.c: Header 6 [ 16]: CSeq: 102 INVITE [Nov 4 20:18:23] DEBUG[3503] chan_sip.c: Header 7 [ 32]: User-Agent: 3CXPhone 4.0.10858.0 [Nov 4 20:18:23] DEBUG[3503] chan_sip.c: Header 8 [ 17]: Content-Length: 0 [Nov 4 20:18:23] VERBOSE[3503] chan_sip.c: --- (9 headers 0 lines) --- [Nov 4 20:18:23] DEBUG[3503] chan_sip.c: = Looking for Call ID: 70fcefbd1612892f47d24c6229e92e83@192.168.10.181:5060 (Checking To) --From tag as34507cfd --To-tag c2778968 [Nov 4 20:18:23] DEBUG[3503] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '70fcefbd1612892f47d24c6229e92e83@192.168.10.181:5060' Request 102: Found [Nov 4 20:18:23] DEBUG[3503] chan_sip.c: SIP response 180 to standard invite [Nov 4 20:18:23] VERBOSE[3585] app_queue.c: -- SIP/671-00000011 is ringing [Nov 4 20:18:23] VERBOSE[3503] chan_sip.c: <--- SIP read from UDP:192.168.10.154:5061 ---> NOTIFY sip:192.168.10.181:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.154:5061;branch=z9hG4bK-74658dcb From: "gj linksys" ;tag=4bebcde1e3145adfo0 To: Call-ID: 31383749-3d40c097@192.168.10.154 CSeq: 33 NOTIFY Max-Forwards: 70 Event: keep-alive User-Agent: Linksys/SPA922-5.2.8 Content-Length: 0 <-------------> [Nov 4 20:18:23] DEBUG[3503] chan_sip.c: Header 0 [ 38]: NOTIFY sip:192.168.10.181:5060 SIP/2.0 [Nov 4 20:18:23] DEBUG[3503] chan_sip.c: Header 1 [ 60]: Via: SIP/2.0/UDP 192.168.10.154:5061;branch=z9hG4bK-74658dcb [Nov 4 20:18:23] DEBUG[3503] chan_sip.c: Header 2 [ 66]: From: "gj linksys" ;tag=4bebcde1e3145adfo0 [Nov 4 20:18:23] DEBUG[3503] chan_sip.c: Header 3 [ 24]: To: [Nov 4 20:18:23] DEBUG[3503] chan_sip.c: Header 4 [ 41]: Call-ID: 31383749-3d40c097@192.168.10.154 [Nov 4 20:18:23] DEBUG[3503] chan_sip.c: Header 5 [ 15]: CSeq: 33 NOTIFY [Nov 4 20:18:23] DEBUG[3503] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Nov 4 20:18:23] DEBUG[3503] chan_sip.c: Header 7 [ 17]: Event: keep-alive [Nov 4 20:18:23] DEBUG[3503] chan_sip.c: Header 8 [ 32]: User-Agent: Linksys/SPA922-5.2.8 [Nov 4 20:18:23] DEBUG[3503] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Nov 4 20:18:23] VERBOSE[3503] chan_sip.c: --- (10 headers 0 lines) --- [Nov 4 20:18:23] DEBUG[3503] chan_sip.c: = Looking for Call ID: 31383749-3d40c097@192.168.10.154 (Checking From) --From tag 4bebcde1e3145adfo0 --To-tag [Nov 4 20:18:23] DEBUG[3503] chan_sip.c: **** Received NOTIFY (4) - Command in SIP NOTIFY [Nov 4 20:18:23] DEBUG[3503] chan_sip.c: Got NOTIFY Event: keep-alive [Nov 4 20:18:23] VERBOSE[3503] chan_sip.c: <--- Transmitting (no NAT) to 192.168.10.154:5061 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.154:5061;branch=z9hG4bK-74658dcb;received=192.168.10.154 From: "gj linksys" ;tag=4bebcde1e3145adfo0 To: ;tag=as23c1a254 Call-ID: 31383749-3d40c097@192.168.10.154 CSeq: 33 NOTIFY Server: Asterisk PBX 1.8.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> [Nov 4 20:18:23] DEBUG[3503] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.10.154:5061 [Nov 4 20:18:23] VERBOSE[3503] chan_sip.c: Scheduling destruction of SIP dialog '31383749-3d40c097@192.168.10.154' in 32000 ms (Method: NOTIFY) [Nov 4 20:18:23] NOTICE[3503] chan_sip.c: -- Re-registration for gertjan-73@sip.voipcheap.com [Nov 4 20:18:23] VERBOSE[3503] dnsmgr.c: > doing dnsmgr_lookup for 'sip.voipcheap.com' [Nov 4 20:18:23] DEBUG[3503] netsock2.c: Splitting 'sip.voipcheap.com' gives... [Nov 4 20:18:23] DEBUG[3503] netsock2.c: ...host 'sip.voipcheap.com' and port '(null)'. [Nov 4 20:18:23] DEBUG[3503] acl.c: Multiple addresses. Using the first only [Nov 4 20:18:23] DEBUG[3503] chan_sip.c: Allocating new SIP dialog for 1108ab7106a76d686bcecc551c083539@127.0.1.1 - REGISTER (No RTP) [Nov 4 20:18:23] DEBUG[3503] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Nov 4 20:18:23] DEBUG[3503] chan_sip.c: SIP Registry sip.voipcheap.com: refcount now 3 [Nov 4 20:18:23] DEBUG[3503] acl.c: For destination '194.120.0.198', our source address is '192.168.10.181'. [Nov 4 20:18:23] DEBUG[3503] chan_sip.c: Target address 194.120.0.198:5060 is not local, substituting externaddr [Nov 4 20:18:23] DEBUG[3503] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 85.223.76.245:5060 [Nov 4 20:18:23] DEBUG[3503] chan_sip.c: SIP Registry sip.voipcheap.com: refcount now 4 [Nov 4 20:18:23] DEBUG[3503] chan_sip.c: Scheduled a registration timeout for sip.voipcheap.com id #308 [Nov 4 20:18:23] DEBUG[3503] chan_sip.c: >>> Re-using Auth data for gertjan-73@sip.voipcheap.com [Nov 4 20:18:23] DEBUG[3503] chan_sip.c: Initializing initreq for method REGISTER - callid 1108ab7106a76d686bcecc551c083539@127.0.1.1 [Nov 4 20:18:23] DEBUG[3503] chan_sip.c: Header 0 [ 38]: REGISTER sip:sip.voipcheap.com SIP/2.0 [Nov 4 20:18:23] DEBUG[3503] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 85.223.76.245:5060;branch=z9hG4bK518f3aa8 [Nov 4 20:18:23] DEBUG[3503] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 4 20:18:23] DEBUG[3503] chan_sip.c: Header 3 [ 55]: From: ;tag=as2c3e5c25 [Nov 4 20:18:23] DEBUG[3503] chan_sip.c: Header 4 [ 38]: To: [Nov 4 20:18:23] DEBUG[3503] chan_sip.c: Header 5 [ 51]: Call-ID: 1108ab7106a76d686bcecc551c083539@127.0.1.1 [Nov 4 20:18:23] DEBUG[3503] chan_sip.c: Header 6 [ 18]: CSeq: 110 REGISTER [Nov 4 20:18:23] DEBUG[3503] chan_sip.c: Header 7 [ 30]: User-Agent: Asterisk PBX 1.8.0 [Nov 4 20:18:23] DEBUG[3503] chan_sip.c: Header 8 [179]: Authorization: Digest username="gertjan-73", realm="sip.voipcheap.com", algorithm=MD5, uri="sip:sip.voipcheap.com", nonce="1416709203", response="8d5826e6652788a7b6e49d648bc4cf9d" [Nov 4 20:18:23] DEBUG[3503] chan_sip.c: Header 9 [ 12]: Expires: 120 [Nov 4 20:18:23] DEBUG[3503] chan_sip.c: Header 10 [ 35]: Contact: [Nov 4 20:18:23] VERBOSE[3503] chan_sip.c: REGISTER 11 headers, 0 lines [Nov 4 20:18:23] DEBUG[3503] chan_sip.c: REGISTER attempt 1 to gertjan-73@sip.voipcheap.com [Nov 4 20:18:23] VERBOSE[3503] chan_sip.c: Reliably Transmitting (no NAT) to 194.120.0.198:5060: REGISTER sip:sip.voipcheap.com SIP/2.0 Via: SIP/2.0/UDP 85.223.76.245:5060;branch=z9hG4bK518f3aa8 Max-Forwards: 70 From: ;tag=as2c3e5c25 To: Call-ID: 1108ab7106a76d686bcecc551c083539@127.0.1.1 CSeq: 110 REGISTER User-Agent: Asterisk PBX 1.8.0 Authorization: Digest username="gertjan-73", realm="sip.voipcheap.com", algorithm=MD5, uri="sip:sip.voipcheap.com", nonce="1416709203", response="8d5826e6652788a7b6e49d648bc4cf9d" Expires: 120 Contact: Content-Length: 0 --- [Nov 4 20:18:23] DEBUG[3503] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #309 [Nov 4 20:18:23] DEBUG[3503] chan_sip.c: Trying to put 'REGISTER si' onto UDP socket destined for 194.120.0.198:5060 [Nov 4 20:18:23] DEBUG[3503] chan_sip.c: SIP Registry sip.voipcheap.com: refcount now 3 [Nov 4 20:18:23] VERBOSE[3503] chan_sip.c: <--- SIP read from UDP:194.120.0.198:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 85.223.76.245:5060;branch=z9hG4bK518f3aa8 From: ;tag=as2c3e5c25 To: Contact: sip:194.120.0.198:5060 Call-ID: 1108ab7106a76d686bcecc551c083539@127.0.1.1 CSeq: 110 REGISTER Server: (Very nice Sip Registrar/Proxy Server) Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE WWW-Authenticate: Digest realm="sip.voipcheap.com",nonce="1416814265",algorithm=MD5 Content-Length: 0 <-------------> [Nov 4 20:18:23] DEBUG[3503] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Nov 4 20:18:23] DEBUG[3503] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 85.223.76.245:5060;branch=z9hG4bK518f3aa8 [Nov 4 20:18:23] DEBUG[3503] chan_sip.c: Header 2 [ 55]: From: ;tag=as2c3e5c25 [Nov 4 20:18:23] DEBUG[3503] chan_sip.c: Header 3 [ 38]: To: [Nov 4 20:18:23] DEBUG[3503] chan_sip.c: Header 4 [ 31]: Contact: sip:194.120.0.198:5060 [Nov 4 20:18:23] DEBUG[3503] chan_sip.c: Header 5 [ 51]: Call-ID: 1108ab7106a76d686bcecc551c083539@127.0.1.1 [Nov 4 20:18:23] DEBUG[3503] chan_sip.c: Header 6 [ 18]: CSeq: 110 REGISTER [Nov 4 20:18:23] DEBUG[3503] chan_sip.c: Header 7 [ 46]: Server: (Very nice Sip Registrar/Proxy Server) [Nov 4 20:18:23] DEBUG[3503] chan_sip.c: Header 8 [ 58]: Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE [Nov 4 20:18:23] DEBUG[3503] chan_sip.c: Header 9 [ 83]: WWW-Authenticate: Digest realm="sip.voipcheap.com",nonce="1416814265",algorithm=MD5 [Nov 4 20:18:23] DEBUG[3503] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Nov 4 20:18:23] VERBOSE[3503] chan_sip.c: --- (11 headers 0 lines) --- [Nov 4 20:18:23] DEBUG[3503] chan_sip.c: = Looking for Call ID: 1108ab7106a76d686bcecc551c083539@127.0.1.1 (Checking To) --From tag as2c3e5c25 --To-tag [Nov 4 20:18:23] DEBUG[3503] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #309 [Nov 4 20:18:23] DEBUG[3503] chan_sip.c: Stopping retransmission on '1108ab7106a76d686bcecc551c083539@127.0.1.1' of Request 110: Match Found [Nov 4 20:18:23] VERBOSE[3503] chan_sip.c: Responding to challenge, registration to domain/host name sip.voipcheap.com [Nov 4 20:18:23] VERBOSE[3503] dnsmgr.c: > doing dnsmgr_lookup for 'sip.voipcheap.com' [Nov 4 20:18:23] DEBUG[3503] netsock2.c: Splitting 'sip.voipcheap.com' gives... [Nov 4 20:18:23] DEBUG[3503] netsock2.c: ...host 'sip.voipcheap.com' and port '(null)'. [Nov 4 20:18:23] DEBUG[3503] acl.c: Multiple addresses. Using the first only [Nov 4 20:18:23] DEBUG[3503] chan_sip.c: Initializing already initialized SIP dialog 1108ab7106a76d686bcecc551c083539@127.0.1.1 (presumably reinvite) [Nov 4 20:18:23] DEBUG[3503] chan_sip.c: Header 0 [ 38]: REGISTER sip:sip.voipcheap.com SIP/2.0 [Nov 4 20:18:23] DEBUG[3503] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 85.223.76.245:5060;branch=z9hG4bK56c54d3a [Nov 4 20:18:23] DEBUG[3503] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 4 20:18:23] DEBUG[3503] chan_sip.c: Header 3 [ 55]: From: ;tag=as2dfc8e17 [Nov 4 20:18:23] DEBUG[3503] chan_sip.c: Header 4 [ 38]: To: [Nov 4 20:18:23] DEBUG[3503] chan_sip.c: Header 5 [ 51]: Call-ID: 1108ab7106a76d686bcecc551c083539@127.0.1.1 [Nov 4 20:18:23] DEBUG[3503] chan_sip.c: Header 6 [ 18]: CSeq: 111 REGISTER [Nov 4 20:18:23] DEBUG[3503] chan_sip.c: Header 7 [ 30]: User-Agent: Asterisk PBX 1.8.0 [Nov 4 20:18:23] DEBUG[3503] chan_sip.c: Header 8 [179]: Authorization: Digest username="gertjan-73", realm="sip.voipcheap.com", algorithm=MD5, uri="sip:sip.voipcheap.com", nonce="1416814265", response="3dd5f87b7a48125a63f443bf6f17af6e" [Nov 4 20:18:23] DEBUG[3503] chan_sip.c: Header 9 [ 12]: Expires: 120 [Nov 4 20:18:23] DEBUG[3503] chan_sip.c: Header 10 [ 35]: Contact: [Nov 4 20:18:23] VERBOSE[3503] chan_sip.c: REGISTER 11 headers, 0 lines [Nov 4 20:18:23] DEBUG[3503] chan_sip.c: REGISTER attempt 2 to gertjan-73@sip.voipcheap.com [Nov 4 20:18:23] VERBOSE[3503] chan_sip.c: Reliably Transmitting (no NAT) to 194.120.0.198:5060: REGISTER sip:sip.voipcheap.com SIP/2.0 Via: SIP/2.0/UDP 85.223.76.245:5060;branch=z9hG4bK56c54d3a Max-Forwards: 70 From: ;tag=as2dfc8e17 To: Call-ID: 1108ab7106a76d686bcecc551c083539@127.0.1.1 CSeq: 111 REGISTER User-Agent: Asterisk PBX 1.8.0 Authorization: Digest username="gertjan-73", realm="sip.voipcheap.com", algorithm=MD5, uri="sip:sip.voipcheap.com", nonce="1416814265", response="3dd5f87b7a48125a63f443bf6f17af6e" Expires: 120 Contact: Content-Length: 0 --- [Nov 4 20:18:23] DEBUG[3503] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #310 [Nov 4 20:18:23] DEBUG[3503] chan_sip.c: Trying to put 'REGISTER si' onto UDP socket destined for 194.120.0.198:5060 [Nov 4 20:18:24] VERBOSE[3503] chan_sip.c: <--- SIP read from UDP:194.120.0.198:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 85.223.76.245:5060;branch=z9hG4bK56c54d3a From: ;tag=as2dfc8e17 To: Contact: sip:194.120.0.198:5060 Call-ID: 1108ab7106a76d686bcecc551c083539@127.0.1.1 CSeq: 111 REGISTER Server: (Very nice Sip Registrar/Proxy Server) Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE Content-Length: 0 <-------------> [Nov 4 20:18:24] DEBUG[3503] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Nov 4 20:18:24] DEBUG[3503] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 85.223.76.245:5060;branch=z9hG4bK56c54d3a [Nov 4 20:18:24] DEBUG[3503] chan_sip.c: Header 2 [ 55]: From: ;tag=as2dfc8e17 [Nov 4 20:18:24] DEBUG[3503] chan_sip.c: Header 3 [ 38]: To: [Nov 4 20:18:24] DEBUG[3503] chan_sip.c: Header 4 [ 31]: Contact: sip:194.120.0.198:5060 [Nov 4 20:18:24] DEBUG[3503] chan_sip.c: Header 5 [ 51]: Call-ID: 1108ab7106a76d686bcecc551c083539@127.0.1.1 [Nov 4 20:18:24] DEBUG[3503] chan_sip.c: Header 6 [ 18]: CSeq: 111 REGISTER [Nov 4 20:18:24] DEBUG[3503] chan_sip.c: Header 7 [ 46]: Server: (Very nice Sip Registrar/Proxy Server) [Nov 4 20:18:24] DEBUG[3503] chan_sip.c: Header 8 [ 58]: Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE [Nov 4 20:18:24] DEBUG[3503] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Nov 4 20:18:24] VERBOSE[3503] chan_sip.c: --- (10 headers 0 lines) --- [Nov 4 20:18:24] DEBUG[3503] chan_sip.c: = Looking for Call ID: 1108ab7106a76d686bcecc551c083539@127.0.1.1 (Checking To) --From tag as2dfc8e17 --To-tag [Nov 4 20:18:24] VERBOSE[3503] chan_sip.c: <--- SIP read from UDP:194.120.0.198:5060 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 85.223.76.245:5060;branch=z9hG4bK56c54d3a From: ;tag=as2dfc8e17 To: Contact: ;expires=120 Call-ID: 1108ab7106a76d686bcecc551c083539@127.0.1.1 CSeq: 111 REGISTER Server: (Very nice Sip Registrar/Proxy Server) Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE Content-Length: 0 <-------------> [Nov 4 20:18:24] DEBUG[3503] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 Ok [Nov 4 20:18:24] DEBUG[3503] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 85.223.76.245:5060;branch=z9hG4bK56c54d3a [Nov 4 20:18:24] DEBUG[3503] chan_sip.c: Header 2 [ 55]: From: ;tag=as2dfc8e17 [Nov 4 20:18:24] DEBUG[3503] chan_sip.c: Header 3 [ 38]: To: [Nov 4 20:18:24] DEBUG[3503] chan_sip.c: Header 4 [ 47]: Contact: ;expires=120 [Nov 4 20:18:24] DEBUG[3503] chan_sip.c: Header 5 [ 51]: Call-ID: 1108ab7106a76d686bcecc551c083539@127.0.1.1 [Nov 4 20:18:24] DEBUG[3503] chan_sip.c: Header 6 [ 18]: CSeq: 111 REGISTER [Nov 4 20:18:24] DEBUG[3503] chan_sip.c: Header 7 [ 46]: Server: (Very nice Sip Registrar/Proxy Server) [Nov 4 20:18:24] DEBUG[3503] chan_sip.c: Header 8 [ 58]: Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE [Nov 4 20:18:24] DEBUG[3503] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Nov 4 20:18:24] VERBOSE[3503] chan_sip.c: --- (10 headers 0 lines) --- [Nov 4 20:18:24] DEBUG[3503] chan_sip.c: = Looking for Call ID: 1108ab7106a76d686bcecc551c083539@127.0.1.1 (Checking To) --From tag as2dfc8e17 --To-tag [Nov 4 20:18:24] DEBUG[3503] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #310 [Nov 4 20:18:24] DEBUG[3503] chan_sip.c: Stopping retransmission on '1108ab7106a76d686bcecc551c083539@127.0.1.1' of Request 111: Match Found [Nov 4 20:18:24] DEBUG[3508] manager.c: Examining event: Event: Registry Privilege: system,all ChannelType: SIP Domain: sip.voipcheap.com Status: Registered [Nov 4 20:18:24] DEBUG[3503] chan_sip.c: Registration successful [Nov 4 20:18:24] DEBUG[3503] chan_sip.c: Cancelling timeout 308 [Nov 4 20:18:24] DEBUG[3503] chan_sip.c: SIP Registry sip.voipcheap.com: refcount now 2 [Nov 4 20:18:24] DEBUG[3503] chan_sip.c: SIP Registry sip.voipcheap.com: refcount now 1 [Nov 4 20:18:24] VERBOSE[3503] chan_sip.c: Scheduling destruction of SIP dialog '1108ab7106a76d686bcecc551c083539@127.0.1.1' in 32000 ms (Method: REGISTER) [Nov 4 20:18:24] NOTICE[3503] chan_sip.c: Outbound Registration: Expiry for sip.voipcheap.com is 120 sec (Scheduling reregistration in 105 s) [Nov 4 20:18:24] DEBUG[3503] chan_sip.c: SIP Registry sip.voipcheap.com: refcount now 2 [Nov 4 20:18:27] VERBOSE[3503] chan_sip.c: <--- SIP read from UDP:192.168.10.163:5060 ---> <-------------> [Nov 4 20:18:27] DEBUG[3503] chan_sip.c: Header 0 [ 0]: [Nov 4 20:18:27] DEBUG[3585] res_rtp_asterisk.c: Got RTCP report of 80 bytes [Nov 4 20:18:27] VERBOSE[3503] chan_sip.c: <--- SIP read from UDP:192.168.10.163:5060 ---> BYE sip:500@192.168.10.181:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.163;branch=z9hG4bK85bff56b911374C From: "623" ;tag=FEC92782-26913BA1 To: ;tag=as7394ab0e CSeq: 3 BYE Call-ID: 1cfa11fe-cfdcbec0-546ab0df@192.168.10.163 Contact: User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.2.2.0084 Authorization: Digest username="623", realm="asterisk", nonce="6a08ddaa", uri="sip:500@192.168.10.181;user=phone", response="c9372e83e7e2b34e626ed259bae12ebf", algorithm=MD5 Max-Forwards: 70 Content-Length: 0 <-------------> [Nov 4 20:18:27] DEBUG[3503] chan_sip.c: Header 0 [ 39]: BYE sip:500@192.168.10.181:5060 SIP/2.0 [Nov 4 20:18:27] DEBUG[3503] chan_sip.c: Header 1 [ 61]: Via: SIP/2.0/UDP 192.168.10.163;branch=z9hG4bK85bff56b911374C [Nov 4 20:18:27] DEBUG[3503] chan_sip.c: Header 2 [ 58]: From: "623" ;tag=FEC92782-26913BA1 [Nov 4 20:18:27] DEBUG[3503] chan_sip.c: Header 3 [ 54]: To: ;tag=as7394ab0e [Nov 4 20:18:27] DEBUG[3503] chan_sip.c: Header 4 [ 11]: CSeq: 3 BYE [Nov 4 20:18:27] DEBUG[3503] chan_sip.c: Header 5 [ 50]: Call-ID: 1cfa11fe-cfdcbec0-546ab0df@192.168.10.163 [Nov 4 20:18:27] DEBUG[3503] chan_sip.c: Header 6 [ 33]: Contact: [Nov 4 20:18:27] DEBUG[3503] chan_sip.c: Header 7 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.2.2.0084 [Nov 4 20:18:27] DEBUG[3503] chan_sip.c: Header 8 [173]: Authorization: Digest username="623", realm="asterisk", nonce="6a08ddaa", uri="sip:500@192.168.10.181;user=phone", response="c9372e83e7e2b34e626ed259bae12ebf", algorithm=MD5 [Nov 4 20:18:27] DEBUG[3503] chan_sip.c: Header 9 [ 16]: Max-Forwards: 70 [Nov 4 20:18:27] DEBUG[3503] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Nov 4 20:18:27] VERBOSE[3503] chan_sip.c: --- (11 headers 0 lines) --- [Nov 4 20:18:27] DEBUG[3503] chan_sip.c: = Looking for Call ID: 1cfa11fe-cfdcbec0-546ab0df@192.168.10.163 (Checking From) --From tag FEC92782-26913BA1 --To-tag as7394ab0e [Nov 4 20:18:27] DEBUG[3503] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [Nov 4 20:18:27] DEBUG[3503] chan_sip.c: Initializing initreq for method BYE - callid 1cfa11fe-cfdcbec0-546ab0df@192.168.10.163 [Nov 4 20:18:27] DEBUG[3503] netsock2.c: Splitting '192.168.10.163' gives... [Nov 4 20:18:27] DEBUG[3503] netsock2.c: ...host '192.168.10.163' and port '(null)'. [Nov 4 20:18:27] VERBOSE[3503] chan_sip.c: Sending to 192.168.10.163:5060 (no NAT) [Nov 4 20:18:27] DEBUG[3503] chan_sip.c: Setting SIP_ALREADYGONE on dialog 1cfa11fe-cfdcbec0-546ab0df@192.168.10.163 [Nov 4 20:18:27] DEBUG[3503] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xd320888' [Nov 4 20:18:27] VERBOSE[3503] chan_sip.c: Scheduling destruction of SIP dialog '1cfa11fe-cfdcbec0-546ab0df@192.168.10.163' in 32000 ms (Method: BYE) [Nov 4 20:18:27] DEBUG[3503] chan_sip.c: Received bye, issuing owner hangup [Nov 4 20:18:27] VERBOSE[3503] chan_sip.c: <--- Transmitting (no NAT) to 192.168.10.163:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.163;branch=z9hG4bK85bff56b911374C;received=192.168.10.163 From: "623" ;tag=FEC92782-26913BA1 To: ;tag=as7394ab0e Call-ID: 1cfa11fe-cfdcbec0-546ab0df@192.168.10.163 CSeq: 3 BYE Server: Asterisk PBX 1.8.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> [Nov 4 20:18:27] DEBUG[3503] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.10.163:5060 [Nov 4 20:18:27] VERBOSE[3585] res_musiconhold.c: -- Stopped music on hold on SIP/623-00000010 [Nov 4 20:18:27] DEBUG[3585] channel.c: Set channel SIP/623-00000010 to write format ulaw [Nov 4 20:18:27] DEBUG[3585] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Nov 4 20:18:27] DEBUG[3585] app_queue.c: SIP/623-00000010: Nobody answered. [Nov 4 20:18:27] DEBUG[3585] channel.c: Hanging up channel 'SIP/671-00000011' [Nov 4 20:18:27] DEBUG[3585] chan_sip.c: Hangup call SIP/671-00000011, SIP callid 70fcefbd1612892f47d24c6229e92e83@192.168.10.181:5060 [Nov 4 20:18:27] DEBUG[3585] chan_sip.c: update_call_counter(671) - decrement call limit counter on hangup [Nov 4 20:18:27] DEBUG[3585] chan_sip.c: Updating call counter for outgoing call [Nov 4 20:18:27] DEBUG[3585] chan_sip.c: Call to peer '671' removed from call limit 2 [Nov 4 20:18:27] DEBUG[3495] devicestate.c: No provider found, checking channel drivers for SIP - 671 [Nov 4 20:18:27] DEBUG[3495] chan_sip.c: Checking device state for peer 671 [Nov 4 20:18:27] DEBUG[3495] devicestate.c: Changing state for SIP/671 - state 1 (Not in use) [Nov 4 20:18:27] DEBUG[3495] devicestate.c: device 'SIP/671' state '1' [Nov 4 20:18:27] DEBUG[3533] app_queue.c: Device 'SIP/671' changed to state '1' (Not in use) [Nov 4 20:18:27] DEBUG[3508] manager.c: Examining event: Event: QueueMemberStatus Privilege: agent,all Queue: queue_support Location: SIP/671 MemberName: SIP/671 Membership: static Penalty: 0 CallsTaken: 1 LastCall: 1288898028 Status: 1 Paused: 0 [Nov 4 20:18:27] DEBUG[3585] chan_sip.c: Hanging up channel in state Ringing (not UP) [Nov 4 20:18:27] DEBUG[3585] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xd2c3798' [Nov 4 20:18:27] VERBOSE[3585] chan_sip.c: Scheduling destruction of SIP dialog '70fcefbd1612892f47d24c6229e92e83@192.168.10.181:5060' in 7040 ms (Method: INVITE) [Nov 4 20:18:27] DEBUG[3585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '70fcefbd1612892f47d24c6229e92e83@192.168.10.181:5060' Request 102: Found [Nov 4 20:18:27] VERBOSE[3585] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.10.10:51944: CANCEL sip:671@192.168.10.10:51944;rinstance=29da946aeae5fdb8 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.181:5060;branch=z9hG4bK02310c25 Max-Forwards: 70 From: "MR X" ;tag=as34507cfd To: Call-ID: 70fcefbd1612892f47d24c6229e92e83@192.168.10.181:5060 CSeq: 102 CANCEL User-Agent: Asterisk PBX 1.8.0 Content-Length: 0 --- [Nov 4 20:18:27] DEBUG[3585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #315 [Nov 4 20:18:27] DEBUG[3585] chan_sip.c: Trying to put 'CANCEL sip:' onto UDP socket destined for 192.168.10.10:51944 [Nov 4 20:18:27] VERBOSE[3585] chan_sip.c: Scheduling destruction of SIP dialog '70fcefbd1612892f47d24c6229e92e83@192.168.10.181:5060' in 7040 ms (Method: INVITE) [Nov 4 20:18:27] DEBUG[3508] manager.c: Examining event: Event: Hangup Privilege: call,all Channel: SIP/671-00000011 Uniqueid: 1288898303.17 CallerIDNum: 671 CallerIDName: 3CX Phone Cause: 16 Cause-txt: Normal Clearing [Nov 4 20:18:27] DEBUG[3495] devicestate.c: No provider found, checking channel drivers for SIP - 671 [Nov 4 20:18:27] DEBUG[3495] chan_sip.c: Checking device state for peer 671 [Nov 4 20:18:27] DEBUG[3495] devicestate.c: Changing state for SIP/671 - state 1 (Not in use) [Nov 4 20:18:27] DEBUG[3495] devicestate.c: device 'SIP/671' state '1' [Nov 4 20:18:27] DEBUG[3533] app_queue.c: Device 'SIP/671' changed to state '1' (Not in use) [Nov 4 20:18:27] DEBUG[3508] manager.c: Examining event: Event: QueueMemberStatus Privilege: agent,all Queue: queue_support Location: SIP/671 MemberName: SIP/671 Membership: static Penalty: 0 CallsTaken: 1 LastCall: 1288898028 Status: 1 Paused: 0 [Nov 4 20:18:27] DEBUG[3508] manager.c: Examining event: Event: QueueCallerAbandon Privilege: agent,all Queue: queue_support Uniqueid: 1288898302.16 Position: 1 OriginalPosition: 1 HoldTime: 4 [Nov 4 20:18:27] DEBUG[3585] app_queue.c: Queue 'queue_support' Leave, Channel 'SIP/623-00000010' [Nov 4 20:18:27] DEBUG[3508] manager.c: Examining event: Event: Leave Privilege: call,all Channel: SIP/623-00000010 Queue: queue_support Count: 0 Position: 1 Uniqueid: 1288898302.16 [Nov 4 20:18:27] DEBUG[3585] pbx.c: Spawn extension (do_support,s,4) exited non-zero on 'SIP/623-00000010' [Nov 4 20:18:27] VERBOSE[3585] pbx.c: == Spawn extension (do_support, s, 4) exited non-zero on 'SIP/623-00000010' [Nov 4 20:18:27] DEBUG[3585] channel.c: Soft-Hanging up channel 'SIP/623-00000010' [Nov 4 20:18:27] DEBUG[3585] pbx.c: Launching 'Hangup' [Nov 4 20:18:27] VERBOSE[3585] pbx.c: -- Executing [h@do_support:1] Hangup("SIP/623-00000010", "") in new stack [Nov 4 20:18:27] DEBUG[3585] pbx.c: Spawn extension (do_support,h,1) exited non-zero on 'SIP/623-00000010' [Nov 4 20:18:27] VERBOSE[3585] pbx.c: == Spawn extension (do_support, h, 1) exited non-zero on 'SIP/623-00000010' [Nov 4 20:18:27] DEBUG[3585] channel.c: Hanging up channel 'SIP/623-00000010' [Nov 4 20:18:27] DEBUG[3585] chan_sip.c: Hangup call SIP/623-00000010, SIP callid 1cfa11fe-cfdcbec0-546ab0df@192.168.10.163 [Nov 4 20:18:27] DEBUG[3585] chan_sip.c: update_call_counter(623) - decrement call limit counter on hangup [Nov 4 20:18:27] DEBUG[3585] chan_sip.c: Updating call counter for incoming call [Nov 4 20:18:27] DEBUG[3585] chan_sip.c: Call from peer '623' removed from call limit 2 [Nov 4 20:18:27] DEBUG[3495] devicestate.c: No provider found, checking channel drivers for SIP - 623 [Nov 4 20:18:27] DEBUG[3495] chan_sip.c: Checking device state for peer 623 [Nov 4 20:18:27] DEBUG[3495] devicestate.c: Changing state for SIP/623 - state 1 (Not in use) [Nov 4 20:18:27] DEBUG[3495] devicestate.c: device 'SIP/623' state '1' [Nov 4 20:18:27] DEBUG[3533] app_queue.c: Device 'SIP/623' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Nov 4 20:18:27] DEBUG[3585] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xd320888' [Nov 4 20:18:27] DEBUG[3508] manager.c: Examining event: Event: Hangup Privilege: call,all Channel: SIP/623-00000010 Uniqueid: 1288898302.16 CallerIDNum: 12345 CallerIDName: MR X Cause: 16 Cause-txt: Normal Clearing [Nov 4 20:18:27] DEBUG[3495] devicestate.c: No provider found, checking channel drivers for SIP - 623 [Nov 4 20:18:27] DEBUG[3495] chan_sip.c: Checking device state for peer 623 [Nov 4 20:18:27] DEBUG[3495] devicestate.c: Changing state for SIP/623 - state 1 (Not in use) [Nov 4 20:18:27] DEBUG[3495] devicestate.c: device 'SIP/623' state '1' [Nov 4 20:18:27] DEBUG[3533] app_queue.c: Device 'SIP/623' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Nov 4 20:18:27] DEBUG[3503] chan_sip.c: SIP TIMER: Rescheduling retransmission #315 (1) CANCEL - 14 [Nov 4 20:18:27] DEBUG[3503] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 220 ms (t1 110 ms (Retrans id #315)) [Nov 4 20:18:27] VERBOSE[3503] chan_sip.c: Retransmitting #1 (no NAT) to 192.168.10.10:51944: CANCEL sip:671@192.168.10.10:51944;rinstance=29da946aeae5fdb8 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.181:5060;branch=z9hG4bK02310c25 Max-Forwards: 70 From: "MR X" ;tag=as34507cfd To: Call-ID: 70fcefbd1612892f47d24c6229e92e83@192.168.10.181:5060 CSeq: 102 CANCEL User-Agent: Asterisk PBX 1.8.0 Content-Length: 0 --- [Nov 4 20:18:27] DEBUG[3503] chan_sip.c: Trying to put 'CANCEL sip:' onto UDP socket destined for 192.168.10.10:51944 [Nov 4 20:18:27] VERBOSE[3503] chan_sip.c: <--- SIP read from UDP:192.168.10.10:51944 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.181:5060;branch=z9hG4bK02310c25 Contact: To: ;tag=c2778968 From: "MR X";tag=as34507cfd Call-ID: 70fcefbd1612892f47d24c6229e92e83@192.168.10.181:5060 CSeq: 102 CANCEL User-Agent: 3CXPhone 4.0.10858.0 Content-Length: 0 <-------------> [Nov 4 20:18:27] DEBUG[3503] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Nov 4 20:18:27] DEBUG[3503] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/UDP 192.168.10.181:5060;branch=z9hG4bK02310c25 [Nov 4 20:18:27] DEBUG[3503] chan_sip.c: Header 2 [ 65]: Contact: [Nov 4 20:18:27] DEBUG[3503] chan_sip.c: Header 3 [ 73]: To: ;tag=c2778968 [Nov 4 20:18:27] DEBUG[3503] chan_sip.c: Header 4 [ 53]: From: "MR X";tag=as34507cfd [Nov 4 20:18:27] DEBUG[3503] chan_sip.c: Header 5 [ 61]: Call-ID: 70fcefbd1612892f47d24c6229e92e83@192.168.10.181:5060 [Nov 4 20:18:27] DEBUG[3503] chan_sip.c: Header 6 [ 16]: CSeq: 102 CANCEL [Nov 4 20:18:27] DEBUG[3503] chan_sip.c: Header 7 [ 32]: User-Agent: 3CXPhone 4.0.10858.0 [Nov 4 20:18:27] DEBUG[3503] chan_sip.c: Header 8 [ 17]: Content-Length: 0 [Nov 4 20:18:27] VERBOSE[3503] chan_sip.c: --- (9 headers 0 lines) --- [Nov 4 20:18:27] DEBUG[3503] chan_sip.c: = Looking for Call ID: 70fcefbd1612892f47d24c6229e92e83@192.168.10.181:5060 (Checking To) --From tag as34507cfd --To-tag c2778968 [Nov 4 20:18:27] DEBUG[3503] chan_sip.c: Acked pending invite 102 [Nov 4 20:18:27] DEBUG[3503] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #315 [Nov 4 20:18:27] DEBUG[3503] chan_sip.c: Stopping retransmission on '70fcefbd1612892f47d24c6229e92e83@192.168.10.181:5060' of Request 102: Match Found [Nov 4 20:18:27] VERBOSE[3503] chan_sip.c: <--- SIP read from UDP:192.168.10.10:51944 ---> SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 192.168.10.181:5060;branch=z9hG4bK02310c25 To: ;tag=c2778968 From: "MR X";tag=as34507cfd Call-ID: 70fcefbd1612892f47d24c6229e92e83@192.168.10.181:5060 CSeq: 102 INVITE User-Agent: 3CXPhone 4.0.10858.0 Content-Length: 0 <-------------> [Nov 4 20:18:27] DEBUG[3503] chan_sip.c: Header 0 [ 30]: SIP/2.0 487 Request Terminated [Nov 4 20:18:27] DEBUG[3503] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/UDP 192.168.10.181:5060;branch=z9hG4bK02310c25 [Nov 4 20:18:27] DEBUG[3503] chan_sip.c: Header 2 [ 73]: To: ;tag=c2778968 [Nov 4 20:18:27] DEBUG[3503] chan_sip.c: Header 3 [ 53]: From: "MR X";tag=as34507cfd [Nov 4 20:18:27] DEBUG[3503] chan_sip.c: Header 4 [ 61]: Call-ID: 70fcefbd1612892f47d24c6229e92e83@192.168.10.181:5060 [Nov 4 20:18:27] DEBUG[3503] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Nov 4 20:18:27] DEBUG[3503] chan_sip.c: Header 6 [ 32]: User-Agent: 3CXPhone 4.0.10858.0 [Nov 4 20:18:27] DEBUG[3503] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [Nov 4 20:18:27] VERBOSE[3503] chan_sip.c: --- (8 headers 0 lines) --- [Nov 4 20:18:27] DEBUG[3503] chan_sip.c: = Looking for Call ID: 70fcefbd1612892f47d24c6229e92e83@192.168.10.181:5060 (Checking To) --From tag as34507cfd --To-tag c2778968 [Nov 4 20:18:27] DEBUG[3503] chan_sip.c: Stopping retransmission on '70fcefbd1612892f47d24c6229e92e83@192.168.10.181:5060' of Request 102: Match Found [Nov 4 20:18:27] DEBUG[3503] chan_sip.c: SIP response 487 to standard invite [Nov 4 20:18:27] VERBOSE[3503] chan_sip.c: Transmitting (no NAT) to 192.168.10.10:51944: ACK sip:671@192.168.10.10:51944;rinstance=29da946aeae5fdb8 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.181:5060;branch=z9hG4bK02310c25 Max-Forwards: 70 From: "MR X" ;tag=as34507cfd To: ;tag=c2778968 Contact: Call-ID: 70fcefbd1612892f47d24c6229e92e83@192.168.10.181:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 1.8.0 Content-Length: 0 --- [Nov 4 20:18:27] DEBUG[3503] chan_sip.c: Trying to put 'ACK sip:671' onto UDP socket destined for 192.168.10.10:51944 [Nov 4 20:18:27] DEBUG[3503] chan_sip.c: Updating call counter for outgoing call [Nov 4 20:18:27] DEBUG[3503] chan_sip.c: Call to peer '671' removed from call limit 2 [Nov 4 20:18:27] DEBUG[3495] devicestate.c: No provider found, checking channel drivers for SIP - 671 [Nov 4 20:18:27] DEBUG[3495] chan_sip.c: Checking device state for peer 671 [Nov 4 20:18:27] DEBUG[3495] devicestate.c: Changing state for SIP/671 - state 1 (Not in use) [Nov 4 20:18:27] DEBUG[3495] devicestate.c: device 'SIP/671' state '1' [Nov 4 20:18:27] DEBUG[3533] app_queue.c: Device 'SIP/671' changed to state '1' (Not in use) [Nov 4 20:18:27] DEBUG[3508] manager.c: Examining event: Event: QueueMemberStatus Privilege: agent,all Queue: queue_support Location: SIP/671 MemberName: SIP/671 Membership: static Penalty: 0 CallsTaken: 1 LastCall: 1288898028 Status: 1 Paused: 0 [Nov 4 20:18:27] DEBUG[3503] chan_sip.c: Setting SIP_ALREADYGONE on dialog 70fcefbd1612892f47d24c6229e92e83@192.168.10.181:5060 [Nov 4 20:18:27] DEBUG[3503] chan_sip.c: Destroying SIP dialog 70fcefbd1612892f47d24c6229e92e83@192.168.10.181:5060 [Nov 4 20:18:27] VERBOSE[3503] chan_sip.c: Really destroying SIP dialog '70fcefbd1612892f47d24c6229e92e83@192.168.10.181:5060' Method: INVITE [Nov 4 20:18:27] DEBUG[3503] rtp_engine.c: Destroyed RTP instance '0xd2c3798' [Nov 4 20:18:28] VERBOSE[3503] chan_sip.c: <--- SIP read from UDP:192.168.10.10:51944 ---> SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP 192.168.10.181:5060;branch=z9hG4bK02310c25 To: ;tag=633e7c2e From: "MR X";tag=as34507cfd Call-ID: 70fcefbd1612892f47d24c6229e92e83@192.168.10.181:5060 CSeq: 102 CANCEL Content-Length: 0 <-------------> [Nov 4 20:18:28] DEBUG[3503] chan_sip.c: Header 0 [ 43]: SIP/2.0 481 Call/Transaction Does Not Exist [Nov 4 20:18:28] DEBUG[3503] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/UDP 192.168.10.181:5060;branch=z9hG4bK02310c25 [Nov 4 20:18:28] DEBUG[3503] chan_sip.c: Header 2 [ 73]: To: ;tag=633e7c2e [Nov 4 20:18:28] DEBUG[3503] chan_sip.c: Header 3 [ 53]: From: "MR X";tag=as34507cfd [Nov 4 20:18:28] DEBUG[3503] chan_sip.c: Header 4 [ 61]: Call-ID: 70fcefbd1612892f47d24c6229e92e83@192.168.10.181:5060 [Nov 4 20:18:28] DEBUG[3503] chan_sip.c: Header 5 [ 16]: CSeq: 102 CANCEL [Nov 4 20:18:28] DEBUG[3503] chan_sip.c: Header 6 [ 17]: Content-Length: 0 [Nov 4 20:18:28] VERBOSE[3503] chan_sip.c: --- (7 headers 0 lines) --- [Nov 4 20:18:28] DEBUG[3503] chan_sip.c: = Looking for Call ID: 70fcefbd1612892f47d24c6229e92e83@192.168.10.181:5060 (Checking To) --From tag as34507cfd --To-tag 633e7c2e [Nov 4 20:18:28] DEBUG[3503] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 70fcefbd1612892f47d24c6229e92e83@192.168.10.181:5060 [Nov 4 20:18:28] DEBUG[3503] chan_sip.c: Invalid SIP message - rejected , no callid, len 329 [Nov 4 20:18:32] VERBOSE[3503] chan_sip.c: <--- SIP read from UDP:192.168.10.163:5060 ---> <-------------> [Nov 4 20:18:32] DEBUG[3503] chan_sip.c: Header 0 [ 0]: [Nov 4 20:18:33] VERBOSE[3503] chan_sip.c: <--- SIP read from UDP:192.168.10.10:51944 ---> <-------------> [Nov 4 20:18:33] DEBUG[3503] chan_sip.c: Header 0 [ 0]: [Nov 4 20:18:36] DEBUG[3503] chan_sip.c: Auto destroying SIP dialog '19184f42-1f5ef404-96d8ea23@192.168.10.163' [Nov 4 20:18:36] DEBUG[3503] chan_sip.c: Destroying SIP dialog 19184f42-1f5ef404-96d8ea23@192.168.10.163 [Nov 4 20:18:38] DEBUG[3503] chan_sip.c: = Looking for Call ID: 31383749-3d40c097@192.168.10.154 (Checking From) --From tag 4bebcde1e3145adfo0 --To-tag [Nov 4 20:18:38] DEBUG[3503] chan_sip.c: **** Received NOTIFY (4) - Command in SIP NOTIFY [Nov 4 20:18:38] DEBUG[3503] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.10.154:5061 [Nov 4 20:18:38] DEBUG[3503] chan_sip.c: Allocating new SIP dialog for 2483a00643f137700541639846d559a5@127.0.1.1:0 - OPTIONS (No RTP) [Nov 4 20:18:38] DEBUG[3503] acl.c: For destination '192.168.10.154', our source address is '192.168.10.181'. [Nov 4 20:18:38] DEBUG[3503] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.10.181:5060 [Nov 4 20:18:38] DEBUG[3503] chan_sip.c: Initializing initreq for method OPTIONS - callid 347308901f6f1a5473e95d1d3f8c01a3@192.168.10.181:5060 [Nov 4 20:18:38] DEBUG[3503] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.10.154:5061 [Nov 4 20:18:38] DEBUG[3503] chan_sip.c: = Looking for Call ID: 347308901f6f1a5473e95d1d3f8c01a3@192.168.10.181:5060 (Checking To) --From tag as6bed87bd --To-tag 793ce3892a0a0157i0 [Nov 4 20:18:38] DEBUG[3503] chan_sip.c: Stopping retransmission on '347308901f6f1a5473e95d1d3f8c01a3@192.168.10.181:5060' of Request 102: Match Found [Nov 4 20:18:38] DEBUG[3503] chan_sip.c: Destroying SIP dialog 347308901f6f1a5473e95d1d3f8c01a3@192.168.10.181:5060 [Nov 4 20:18:39] DEBUG[3503] chan_sip.c: Allocating new SIP dialog for 24d15b10360333cf3bb0986d145ad161@127.0.1.1:0 - OPTIONS (No RTP) [Nov 4 20:18:39] DEBUG[3503] acl.c: For destination '83.143.188.165', our source address is '192.168.10.181'. [Nov 4 20:18:39] DEBUG[3503] chan_sip.c: Target address 83.143.188.165:5060 is not local, substituting externaddr [Nov 4 20:18:39] DEBUG[3503] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 85.223.76.245:5060 [Nov 4 20:18:39] DEBUG[3503] chan_sip.c: Initializing initreq for method OPTIONS - callid 3c869da67901e6c96deda8761db2a4e1@85.223.76.245:5060 [Nov 4 20:18:39] DEBUG[3503] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 83.143.188.165:5060 [Nov 4 20:18:39] DEBUG[3503] chan_sip.c: = Looking for Call ID: 3c869da67901e6c96deda8761db2a4e1@85.223.76.245:5060 (Checking To) --From tag as1590dfaa --To-tag 91003fdeb24a2cb33b34489016d6cf2c.76d3 [Nov 4 20:18:39] DEBUG[3503] chan_sip.c: Stopping retransmission on '3c869da67901e6c96deda8761db2a4e1@85.223.76.245:5060' of Request 102: Match Found [Nov 4 20:18:39] DEBUG[3503] chan_sip.c: Destroying SIP dialog 3c869da67901e6c96deda8761db2a4e1@85.223.76.245:5060 [Nov 4 20:18:39] DEBUG[3503] chan_sip.c: Allocating new SIP dialog for 07640b3651bb6e9f7697edd02860827f@127.0.1.1:0 - OPTIONS (No RTP) [Nov 4 20:18:39] DEBUG[3503] acl.c: For destination '194.120.0.198', our source address is '192.168.10.181'. [Nov 4 20:18:39] DEBUG[3503] chan_sip.c: Target address 194.120.0.198:5060 is not local, substituting externaddr [Nov 4 20:18:39] DEBUG[3503] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 85.223.76.245:5060 [Nov 4 20:18:39] DEBUG[3503] chan_sip.c: Initializing initreq for method OPTIONS - callid 7f4a1e4b36cb73ac798f070c3f02c745@85.223.76.245:5060 [Nov 4 20:18:39] DEBUG[3503] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 194.120.0.198:5060 [Nov 4 20:18:39] DEBUG[3503] chan_sip.c: = Looking for Call ID: 7f4a1e4b36cb73ac798f070c3f02c745@85.223.76.245:5060 (Checking To) --From tag as75f10d92 --To-tag [Nov 4 20:18:39] DEBUG[3503] chan_sip.c: Stopping retransmission on '7f4a1e4b36cb73ac798f070c3f02c745@85.223.76.245:5060' of Request 102: Match Found [Nov 4 20:18:39] DEBUG[3503] chan_sip.c: Destroying SIP dialog 7f4a1e4b36cb73ac798f070c3f02c745@85.223.76.245:5060 [Nov 4 20:18:39] DEBUG[3503] chan_sip.c: Allocating new SIP dialog for 02436d1e4bfba9351d8d96626947fc25@127.0.1.1:0 - OPTIONS (No RTP) [Nov 4 20:18:39] DEBUG[3503] acl.c: For destination '192.168.10.10', our source address is '192.168.10.181'. [Nov 4 20:18:39] DEBUG[3503] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.10.181:5060 [Nov 4 20:18:39] DEBUG[3503] chan_sip.c: Initializing initreq for method OPTIONS - callid 06f8ded726233c5b6f602a384669c1bc@192.168.10.181:5060 [Nov 4 20:18:39] DEBUG[3503] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.10.10:51944 [Nov 4 20:18:39] DEBUG[3503] chan_sip.c: Allocating new SIP dialog for 43d5b06353467883436e6aa81f490aed@127.0.1.1:0 - OPTIONS (No RTP) [Nov 4 20:18:39] DEBUG[3503] acl.c: For destination '77.72.169.131', our source address is '192.168.10.181'. [Nov 4 20:18:39] DEBUG[3503] chan_sip.c: Target address 77.72.169.131:5060 is not local, substituting externaddr [Nov 4 20:18:39] DEBUG[3503] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 85.223.76.245:5060 [Nov 4 20:18:39] DEBUG[3503] chan_sip.c: Initializing initreq for method OPTIONS - callid 5400edb34871464c05bbb8e57b318650@85.223.76.245:5060 [Nov 4 20:18:39] DEBUG[3503] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 77.72.169.131:5060 [Nov 4 20:18:39] DEBUG[3503] chan_sip.c: = Looking for Call ID: 5400edb34871464c05bbb8e57b318650@85.223.76.245:5060 (Checking To) --From tag as630d2ac1 --To-tag [Nov 4 20:18:39] DEBUG[3503] chan_sip.c: Stopping retransmission on '5400edb34871464c05bbb8e57b318650@85.223.76.245:5060' of Request 102: Match Found [Nov 4 20:18:39] DEBUG[3503] chan_sip.c: Destroying SIP dialog 5400edb34871464c05bbb8e57b318650@85.223.76.245:5060 [Nov 4 20:18:39] DEBUG[3503] chan_sip.c: = Looking for Call ID: 06f8ded726233c5b6f602a384669c1bc@192.168.10.181:5060 (Checking To) --From tag as0c02d9e5 --To-tag c11f571d [Nov 4 20:18:39] DEBUG[3503] chan_sip.c: Stopping retransmission on '06f8ded726233c5b6f602a384669c1bc@192.168.10.181:5060' of Request 102: Match Found [Nov 4 20:18:39] DEBUG[3503] chan_sip.c: Destroying SIP dialog 06f8ded726233c5b6f602a384669c1bc@192.168.10.181:5060