[2009-11-13 15:36:42.832] VERBOSE[15878] chan_sip.c: <--- SIP read from UDP://10.0.1.10:5060 ---> INVITE sip:5555@10.0.1.2 SIP/2.0 Via: SIP/2.0/UDP 10.0.1.10:5060;branch=z9hG4bK-5791164-7682 From: "07711223344" ;tag=579116425133 To: Call-ID: 579116414152@10.0.1.10 CSeq: 20 INVITE Contact: Max-Forwards: 70 User-Agent: iS3000 SIP Server Privacy: none P-Asserted-Identity: "07711223344" P-Asserted-Identity: Alert-Info: Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, REFER, SUBSCRIBE Content-Type: application/sdp Accept: application/sdp Content-Length: 251 v=0 o=iS3000 20 20 IN IP4 10.0.1.11 s=- c=IN IP4 10.0.1.11 t=0 0 m=audio 49348 RTP/AVP 8 0 18 0 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 telephone-event/8000 a=ptime:30 a=sendrecv <-------------> [2009-11-13 15:36:42.832] DEBUG[15878] chan_sip.c: Header 0 [ 37]: INVITE sip:5555@10.0.1.2 SIP/2.0 [2009-11-13 15:36:42.832] DEBUG[15878] chan_sip.c: Header 1 [ 61]: Via: SIP/2.0/UDP 10.0.1.10:5060;branch=z9hG4bK-5791164-7682 [2009-11-13 15:36:42.833] DEBUG[15878] chan_sip.c: Header 2 [ 64]: From: "07711223344" ;tag=579116425133 [2009-11-13 15:36:42.833] DEBUG[15878] chan_sip.c: Header 3 [ 28]: To: [2009-11-13 15:36:42.833] DEBUG[15878] chan_sip.c: Header 4 [ 33]: Call-ID: 579116414152@10.0.1.10 [2009-11-13 15:36:42.833] DEBUG[15878] chan_sip.c: Header 5 [ 15]: CSeq: 20 INVITE [2009-11-13 15:36:42.833] DEBUG[15878] chan_sip.c: Header 6 [ 42]: Contact: [2009-11-13 15:36:42.833] DEBUG[15878] chan_sip.c: Header 7 [ 16]: Max-Forwards: 70 [2009-11-13 15:36:42.833] DEBUG[15878] chan_sip.c: Header 8 [ 29]: User-Agent: iS3000 SIP Server [2009-11-13 15:36:42.833] DEBUG[15878] chan_sip.c: Header 9 [ 13]: Privacy: none [2009-11-13 15:36:42.833] DEBUG[15878] chan_sip.c: Header 10 [ 62]: P-Asserted-Identity: "07711223344" [2009-11-13 15:36:42.833] DEBUG[15878] chan_sip.c: Header 11 [ 37]: P-Asserted-Identity: [2009-11-13 15:36:42.833] DEBUG[15878] chan_sip.c: Header 12 [ 27]: Alert-Info: [2009-11-13 15:36:42.833] DEBUG[15878] chan_sip.c: Header 13 [ 58]: Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, REFER, SUBSCRIBE [2009-11-13 15:36:42.833] DEBUG[15878] chan_sip.c: Header 14 [ 29]: Content-Type: application/sdp [2009-11-13 15:36:42.833] DEBUG[15878] chan_sip.c: Header 15 [ 23]: Accept: application/sdp [2009-11-13 15:36:42.833] DEBUG[15878] chan_sip.c: Header 16 [ 21]: Content-Length: 251 [2009-11-13 15:36:42.833] DEBUG[15878] chan_sip.c: Header 17 [ 0]: [2009-11-13 15:36:42.833] DEBUG[15878] chan_sip.c: Body 0 [ 3]: v=0 [2009-11-13 15:36:42.833] DEBUG[15878] chan_sip.c: Body 1 [ 33]: o=iS3000 20 20 IN IP4 10.0.1.11 [2009-11-13 15:36:42.833] DEBUG[15878] chan_sip.c: Body 2 [ 3]: s=- [2009-11-13 15:36:42.833] DEBUG[15878] chan_sip.c: Body 3 [ 20]: c=IN IP4 10.0.1.11 [2009-11-13 15:36:42.833] DEBUG[15878] chan_sip.c: Body 4 [ 5]: t=0 0 [2009-11-13 15:36:42.833] DEBUG[15878] chan_sip.c: Body 5 [ 30]: m=audio 49348 RTP/AVP 8 0 18 0 [2009-11-13 15:36:42.833] DEBUG[15878] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [2009-11-13 15:36:42.833] DEBUG[15878] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [2009-11-13 15:36:42.833] DEBUG[15878] chan_sip.c: Body 8 [ 21]: a=rtpmap:18 G729/8000 [2009-11-13 15:36:42.833] DEBUG[15878] chan_sip.c: Body 9 [ 19]: a=fmtp:18 annexb=no [2009-11-13 15:36:42.833] DEBUG[15878] chan_sip.c: Body 10 [ 31]: a=rtpmap:0 telephone-event/8000 [2009-11-13 15:36:42.833] DEBUG[15878] chan_sip.c: Body 11 [ 10]: a=ptime:30 [2009-11-13 15:36:42.833] DEBUG[15878] chan_sip.c: Body 12 [ 10]: a=sendrecv [2009-11-13 15:36:42.833] VERBOSE[15878] chan_sip.c: --- (17 headers 13 lines) --- [2009-11-13 15:36:42.833] DEBUG[15878] acl.c: Found IP address for this socket [2009-11-13 15:36:42.833] VERBOSE[15878] netsock.c: == Using SIP RTP CoS mark 5 [2009-11-13 15:36:42.833] DEBUG[15878] chan_sip.c: Setting NAT on RTP to Off [2009-11-13 15:36:42.834] DEBUG[15878] chan_sip.c: Allocating new SIP dialog for 579116414152@10.0.1.10 - INVITE (With RTP) [2009-11-13 15:36:42.834] DEBUG[15878] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [2009-11-13 15:36:42.834] VERBOSE[15878] chan_sip.c: Sending to 10.0.1.10 : 5060 (no NAT) [2009-11-13 15:36:42.834] DEBUG[15878] chan_sip.c: Initializing initreq for method INVITE - callid 579116414152@10.0.1.10 [2009-11-13 15:36:42.834] VERBOSE[15878] chan_sip.c: Using INVITE request as basis request - 579116414152@10.0.1.10 [2009-11-13 15:36:42.834] VERBOSE[15878] chan_sip.c: Found peer 'siptrunk1' for '07711223344' from 10.0.1.10:5060 [2009-11-13 15:36:42.834] DEBUG[15878] chan_sip.c: Setting NAT on RTP to Off [2009-11-13 15:36:42.834] VERBOSE[15878] chan_sip.c: Found RTP audio format 8 [2009-11-13 15:36:42.834] VERBOSE[15878] chan_sip.c: Found RTP audio format 0 [2009-11-13 15:36:42.834] VERBOSE[15878] chan_sip.c: Found RTP audio format 18 [2009-11-13 15:36:42.834] VERBOSE[15878] chan_sip.c: Found RTP audio format 0 [2009-11-13 15:36:42.834] VERBOSE[15878] chan_sip.c: Peer audio RTP is at port 10.0.1.11:49348 [2009-11-13 15:36:42.834] VERBOSE[15878] chan_sip.c: Found audio description format PCMA for ID 8 [2009-11-13 15:36:42.834] VERBOSE[15878] chan_sip.c: Found audio description format PCMU for ID 0 [2009-11-13 15:36:42.834] VERBOSE[15878] chan_sip.c: Found audio description format G729 for ID 18 [2009-11-13 15:36:42.834] VERBOSE[15878] chan_sip.c: Got unsupported a:fmtp in SDP offer [2009-11-13 15:36:42.834] VERBOSE[15878] chan_sip.c: Found audio description format telephone-event for ID 0 [2009-11-13 15:36:42.834] VERBOSE[15878] chan_sip.c: Capabilities: us - 0x8 (alaw), peer - audio=0x108 (alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) [2009-11-13 15:36:42.834] VERBOSE[15878] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [2009-11-13 15:36:42.834] VERBOSE[15878] chan_sip.c: Peer audio RTP is at port 10.0.1.11:49348 [2009-11-13 15:36:42.834] DEBUG[15878] chan_sip.c: We're settling with these formats: 0x8 (alaw) [2009-11-13 15:36:42.834] DEBUG[15878] chan_sip.c: Checking SIP call limits for device 5555 [2009-11-13 15:36:42.834] DEBUG[15878] chan_sip.c: Updating call counter for incoming call [2009-11-13 15:36:42.835] DEBUG[15878] chan_sip.c: Call from peer 'siptrunk1' is 1 out of 1000 [2009-11-13 15:36:42.835] VERBOSE[15878] chan_sip.c: Looking for 5555 in default (domain 10.0.1.2) [2009-11-13 15:36:42.835] DEBUG[15878] chan_sip.c: *** Our native formats are 0x8 (alaw) [2009-11-13 15:36:42.835] DEBUG[15878] chan_sip.c: *** Joint capabilities are 0x8 (alaw) [2009-11-13 15:36:42.835] DEBUG[15878] chan_sip.c: *** Our capabilities are 0x8 (alaw) [2009-11-13 15:36:42.835] DEBUG[15878] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x8 (alaw) [2009-11-13 15:36:42.835] DEBUG[15878] chan_sip.c: This channel will not be able to handle video. [2009-11-13 15:36:42.835] DEBUG[15878] chan_sip.c: build_route: Contact hop: [2009-11-13 15:36:42.835] VERBOSE[15878] chan_sip.c: list_route: hop: [2009-11-13 15:36:42.835] DEBUG[15878] chan_sip.c: SIP/siptrunk1-0a2ec2b0: New call is still down.... Trying... [2009-11-13 15:36:42.835] VERBOSE[15878] chan_sip.c: <--- Transmitting (no NAT) to 10.0.1.10:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.0.1.10:5060;branch=z9hG4bK-5791164-7682;received=10.0.1.10 From: "07711223344" ;tag=579116425133 To: Call-ID: 579116414152@10.0.1.10 CSeq: 20 INVITE Server: Asterisk PBX 1.6.1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: Content-Length: 0 <------------> [2009-11-13 15:36:42.835] DEBUG[15878] chan_sip.c: Trying to put 'SIP/2.0 10' onto UDP socket destined for 10.0.1.10:5060 [2009-11-13 15:36:42.837] DEBUG[15901] pbx.c: Launching 'Answer' [2009-11-13 15:36:42.837] VERBOSE[15901] pbx.c: -- Executing [5555@default:1] Answer("SIP/siptrunk1-0a2ec2b0", "") in new stack [2009-11-13 15:36:42.837] DEBUG[15901] chan_sip.c: SIP answering channel: SIP/siptrunk1-0a2ec2b0 [2009-11-13 15:36:42.837] DEBUG[15901] chan_sip.c: Setting framing from config on incoming call [2009-11-13 15:36:42.837] DEBUG[15901] chan_sip.c: ** Our capability: 0x8 (alaw) Video flag: True Text flag: True [2009-11-13 15:36:42.837] DEBUG[15901] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [2009-11-13 15:36:42.837] VERBOSE[15901] chan_sip.c: Audio is at 10.0.1.2 port 19374 [2009-11-13 15:36:42.837] VERBOSE[15901] chan_sip.c: Adding codec 0x8 (alaw) to SDP [2009-11-13 15:36:42.837] VERBOSE[15901] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [2009-11-13 15:36:42.837] DEBUG[15901] chan_sip.c: -- Done with adding codecs to SDP [2009-11-13 15:36:42.837] DEBUG[15901] chan_sip.c: Done building SDP. Settling with this capability: 0x8 (alaw) [2009-11-13 15:36:42.838] VERBOSE[15901] chan_sip.c: <--- Reliably Transmitting (no NAT) to 10.0.1.10:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.1.10:5060;branch=z9hG4bK-5791164-7682;received=10.0.1.10 From: "07711223344" ;tag=579116425133 To: ;tag=as54b68200 Call-ID: 579116414152@10.0.1.10 CSeq: 20 INVITE Server: Asterisk PBX 1.6.1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 242 v=0 o=root 1909576850 1909576850 IN IP4 10.0.1.2 s=Asterisk PBX 1.6.1.0 c=IN IP4 10.0.1.2 t=0 0 m=audio 19374 RTP/AVP 8 0 a=rtpmap:8 PCMA/8000 a=rtpmap:0 telephone-event/8000 a=fmtp:0 0-16 a=ptime:20 a=sendrecv <------------> [2009-11-13 15:36:42.838] DEBUG[15901] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #15 [2009-11-13 15:36:42.838] DEBUG[15901] chan_sip.c: Trying to put 'SIP/2.0 20' onto UDP socket destined for 10.0.1.10:5060 [2009-11-13 15:36:42.838] DEBUG[15867] devicestate.c: No provider found, checking channel drivers for SIP - siptrunk1 [2009-11-13 15:36:42.838] DEBUG[15867] chan_sip.c: Checking device state for peer siptrunk1 [2009-11-13 15:36:42.838] DEBUG[15867] devicestate.c: Changing state for SIP/siptrunk1 - state 2 (In use) [2009-11-13 15:36:42.838] DEBUG[15867] devicestate.c: device 'SIP/siptrunk1' state '2' [2009-11-13 15:36:42.838] DEBUG[15867] devicestate.c: No provider found, checking channel drivers for SIP - siptrunk1 [2009-11-13 15:36:42.838] DEBUG[15867] chan_sip.c: Checking device state for peer siptrunk1 [2009-11-13 15:36:42.839] DEBUG[15867] devicestate.c: Changing state for SIP/siptrunk1 - state 2 (In use) [2009-11-13 15:36:42.839] DEBUG[15867] devicestate.c: device 'SIP/siptrunk1' state '2' [2009-11-13 15:36:42.839] DEBUG[15867] devicestate.c: No provider found, checking channel drivers for SIP - siptrunk1 [2009-11-13 15:36:42.839] DEBUG[15867] chan_sip.c: Checking device state for peer siptrunk1 [2009-11-13 15:36:42.839] DEBUG[15867] devicestate.c: Changing state for SIP/siptrunk1 - state 2 (In use) [2009-11-13 15:36:42.839] DEBUG[15867] devicestate.c: device 'SIP/siptrunk1' state '2' [2009-11-13 15:36:42.839] DEBUG[15875] app_queue.c: Device 'SIP/siptrunk1' changed to state '2' (In use) but we don't care because they're not a member of any queue. [2009-11-13 15:36:42.839] DEBUG[15875] app_queue.c: Device 'SIP/siptrunk1' changed to state '2' (In use) but we don't care because they're not a member of any queue. [2009-11-13 15:36:42.839] DEBUG[15875] app_queue.c: Device 'SIP/siptrunk1' changed to state '2' (In use) but we don't care because they're not a member of any queue. [2009-11-13 15:36:42.921] VERBOSE[15878] chan_sip.c: <--- SIP read from UDP://10.0.1.10:5060 ---> ACK sip:5555@10.0.1.2 SIP/2.0 Via: SIP/2.0/UDP 10.0.1.10:5060;branch=z9hG4bK-5791164-7295 From: "07711223344" ;tag=579116425133 To: ;tag=as54b68200 Call-ID: 579116414152@10.0.1.10 CSeq: 20 ACK Max-Forwards: 70 User-Agent: iS3000 SIP Server Content-Length: 0 <-------------> [2009-11-13 15:36:42.921] DEBUG[15878] chan_sip.c: Header 0 [ 34]: ACK sip:5555@10.0.1.2 SIP/2.0 [2009-11-13 15:36:42.921] DEBUG[15878] chan_sip.c: Header 1 [ 61]: Via: SIP/2.0/UDP 10.0.1.10:5060;branch=z9hG4bK-5791164-7295 [2009-11-13 15:36:42.921] DEBUG[15878] chan_sip.c: Header 2 [ 64]: From: "07711223344" ;tag=579116425133 [2009-11-13 15:36:42.921] DEBUG[15878] chan_sip.c: Header 3 [ 43]: To: ;tag=as54b68200 [2009-11-13 15:36:42.921] DEBUG[15878] chan_sip.c: Header 4 [ 33]: Call-ID: 579116414152@10.0.1.10 [2009-11-13 15:36:42.921] DEBUG[15878] chan_sip.c: Header 5 [ 12]: CSeq: 20 ACK [2009-11-13 15:36:42.921] DEBUG[15878] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [2009-11-13 15:36:42.921] DEBUG[15878] chan_sip.c: Header 7 [ 29]: User-Agent: iS3000 SIP Server [2009-11-13 15:36:42.921] DEBUG[15878] chan_sip.c: Header 8 [ 17]: Content-Length: 0 [2009-11-13 15:36:42.921] DEBUG[15878] chan_sip.c: Header 9 [ 0]: [2009-11-13 15:36:42.921] VERBOSE[15878] chan_sip.c: --- (9 headers 0 lines) --- [2009-11-13 15:36:42.921] DEBUG[15878] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [2009-11-13 15:36:42.921] DEBUG[15878] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #15 [2009-11-13 15:36:42.921] DEBUG[15878] chan_sip.c: Stopping retransmission on '579116414152@10.0.1.10' of Response 20: Match Found [2009-11-13 15:36:42.978] DEBUG[15901] pbx.c: Launching 'Wait' [2009-11-13 15:36:42.978] VERBOSE[15901] pbx.c: -- Executing [5555@default:2] Wait("SIP/siptrunk1-0a2ec2b0", "5") in new stack [2009-11-13 15:36:48.211] DEBUG[15901] pbx.c: Launching 'Transfer' [2009-11-13 15:36:48.211] VERBOSE[15901] pbx.c: -- Executing [5555@default:3] Transfer("SIP/siptrunk1-0a2ec2b0", "SIP/5566@10.0.1.10") in new stack [2009-11-13 15:36:48.211] DEBUG[15901] chan_sip.c: SIP transfer of 579116414152@10.0.1.10 to 5566@10.0.1.10 [2009-11-13 15:36:48.211] DEBUG[15901] chan_sip.c: Strict routing enforced for session 579116414152@10.0.1.10 [2009-11-13 15:36:48.212] VERBOSE[15901] chan_sip.c: set_destination: Parsing for address/port to send to [2009-11-13 15:36:48.212] VERBOSE[15901] chan_sip.c: set_destination: set destination to 10.0.1.10, port 5060 [2009-11-13 15:36:48.212] VERBOSE[15901] chan_sip.c: Reliably Transmitting (no NAT) to 10.0.1.10:5060: REFER sip:07711223344@10.0.1.10:5060 SIP/2.0 Via: SIP/2.0/UDP 10.0.1.2:5060;branch=z9hG4bK7bfd80ff;rport Max-Forwards: 70 From: ;tag=as54b68200 To: "07711223344" ;tag=579116425133 Contact: Call-ID: 579116414152@10.0.1.10 CSeq: 102 REFER User-Agent: Asterisk PBX 1.6.1.0 Refer-To: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Referred-By: --- [2009-11-13 15:36:48.212] DEBUG[15901] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #17 [2009-11-13 15:36:48.212] DEBUG[15901] chan_sip.c: Trying to put 'REFER sip:' onto UDP socket destined for 10.0.1.10:5060 [2009-11-13 15:36:48.212] DEBUG[15901] pbx.c: Launching 'Wait' [2009-11-13 15:36:48.212] VERBOSE[15901] pbx.c: -- Executing [5555@default:4] Wait("SIP/siptrunk1-0a2ec2b0", "60") in new stack [2009-11-13 15:36:48.277] VERBOSE[15878] chan_sip.c: <--- SIP read from UDP://10.0.1.10:5060 ---> SIP/2.0 503 Service Unavailable Via: SIP/2.0/UDP 10.0.1.2:5060;branch=z9hG4bK7bfd80ff;rport=5060 From: ;tag=as54b68200 To: "07711223344" ;tag=579116425133 Call-ID: 579116414152@10.0.1.10 CSeq: 102 REFER Contact: User-Agent: iS3000 SIP Server Content-Length: 0 <-------------> [2009-11-13 15:36:48.277] DEBUG[15878] chan_sip.c: Header 0 [ 31]: SIP/2.0 503 Service Unavailable [2009-11-13 15:36:48.277] DEBUG[15878] chan_sip.c: Header 1 [ 69]: Via: SIP/2.0/UDP 10.0.1.2:5060;branch=z9hG4bK7bfd80ff;rport=5060 [2009-11-13 15:36:48.277] DEBUG[15878] chan_sip.c: Header 2 [ 45]: From: ;tag=as54b68200 [2009-11-13 15:36:48.277] DEBUG[15878] chan_sip.c: Header 3 [ 62]: To: "07711223344" ;tag=579116425133 [2009-11-13 15:36:48.277] DEBUG[15878] chan_sip.c: Header 4 [ 33]: Call-ID: 579116414152@10.0.1.10 [2009-11-13 15:36:48.277] DEBUG[15878] chan_sip.c: Header 5 [ 15]: CSeq: 102 REFER [2009-11-13 15:36:48.277] DEBUG[15878] chan_sip.c: Header 6 [ 42]: Contact: [2009-11-13 15:36:48.277] DEBUG[15878] chan_sip.c: Header 7 [ 29]: User-Agent: iS3000 SIP Server [2009-11-13 15:36:48.277] DEBUG[15878] chan_sip.c: Header 8 [ 17]: Content-Length: 0 [2009-11-13 15:36:48.277] DEBUG[15878] chan_sip.c: Header 9 [ 0]: [2009-11-13 15:36:48.278] VERBOSE[15878] chan_sip.c: --- (9 headers 0 lines) --- [2009-11-13 15:36:48.278] DEBUG[15878] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #17 [2009-11-13 15:36:48.278] DEBUG[15878] chan_sip.c: Stopping retransmission on '579116414152@10.0.1.10' of Request 102: Match Found [2009-11-13 15:36:48.278] VERBOSE[15878] chan_sip.c: SIP Response message for INCOMING dialog REFER arrived [2009-11-13 15:36:48.278] VERBOSE[15878] chan_sip.c: -- Incoming call: Got SIP response 503 "Service Unavailable" back from 10.0.1.10 [2009-11-13 15:37:25.534] VERBOSE[15878] chan_sip.c: <--- SIP read from UDP://10.0.1.10:5060 ---> BYE sip:5555@10.0.1.2 SIP/2.0 Via: SIP/2.0/UDP 10.0.1.10:5060;branch=z9hG4bK-5791207-3989 From: "07711223344" ;tag=579116425133 To: ;tag=as54b68200 Call-ID: 579116414152@10.0.1.10 CSeq: 21 BYE Max-Forwards: 70 User-Agent: iS3000 SIP Server Content-Length: 0 <-------------> [2009-11-13 15:37:25.534] DEBUG[15878] chan_sip.c: Header 0 [ 34]: BYE sip:5555@10.0.1.2 SIP/2.0 [2009-11-13 15:37:25.534] DEBUG[15878] chan_sip.c: Header 1 [ 61]: Via: SIP/2.0/UDP 10.0.1.10:5060;branch=z9hG4bK-5791207-3989 [2009-11-13 15:37:25.534] DEBUG[15878] chan_sip.c: Header 2 [ 64]: From: "07711223344" ;tag=579116425133 [2009-11-13 15:37:25.534] DEBUG[15878] chan_sip.c: Header 3 [ 43]: To: ;tag=as54b68200 [2009-11-13 15:37:25.534] DEBUG[15878] chan_sip.c: Header 4 [ 33]: Call-ID: 579116414152@10.0.1.10 [2009-11-13 15:37:25.534] DEBUG[15878] chan_sip.c: Header 5 [ 12]: CSeq: 21 BYE [2009-11-13 15:37:25.534] DEBUG[15878] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [2009-11-13 15:37:25.534] DEBUG[15878] chan_sip.c: Header 7 [ 29]: User-Agent: iS3000 SIP Server [2009-11-13 15:37:25.534] DEBUG[15878] chan_sip.c: Header 8 [ 17]: Content-Length: 0 [2009-11-13 15:37:25.534] DEBUG[15878] chan_sip.c: Header 9 [ 0]: [2009-11-13 15:37:25.534] VERBOSE[15878] chan_sip.c: --- (9 headers 0 lines) --- [2009-11-13 15:37:25.534] DEBUG[15878] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [2009-11-13 15:37:25.534] DEBUG[15878] chan_sip.c: Initializing initreq for method BYE - callid 579116414152@10.0.1.10 [2009-11-13 15:37:25.534] VERBOSE[15878] chan_sip.c: Sending to 10.0.1.10 : 5060 (no NAT) [2009-11-13 15:37:25.535] DEBUG[15878] chan_sip.c: Setting SIP_ALREADYGONE on dialog 579116414152@10.0.1.10 [2009-11-13 15:37:25.535] DEBUG[15878] chan_sip.c: Received bye, issuing owner hangup [2009-11-13 15:37:25.535] VERBOSE[15878] chan_sip.c: <--- Transmitting (no NAT) to 10.0.1.10:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.1.10:5060;branch=z9hG4bK-5791207-3989;received=10.0.1.10 From: "07711223344" ;tag=579116425133 To: ;tag=as54b68200 Call-ID: 579116414152@10.0.1.10 CSeq: 21 BYE Server: Asterisk PBX 1.6.1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Length: 0 <------------> [2009-11-13 15:37:25.535] DEBUG[15878] chan_sip.c: Trying to put 'SIP/2.0 20' onto UDP socket destined for 10.0.1.10:5060 [2009-11-13 15:37:25.535] DEBUG[15901] pbx.c: Spawn extension (default,5555,4) exited non-zero on 'SIP/siptrunk1-0a2ec2b0' [2009-11-13 15:37:25.535] VERBOSE[15901] pbx.c: == Spawn extension (default, 5555, 4) exited non-zero on 'SIP/siptrunk1-0a2ec2b0' [2009-11-13 15:37:25.535] DEBUG[15901] channel.c: Soft-Hanging up channel 'SIP/siptrunk1-0a2ec2b0' [2009-11-13 15:37:25.535] DEBUG[15901] channel.c: Hanging up channel 'SIP/siptrunk1-0a2ec2b0' [2009-11-13 15:37:25.535] DEBUG[15901] chan_sip.c: Hangup call SIP/siptrunk1-0a2ec2b0, SIP callid 579116414152@10.0.1.10 [2009-11-13 15:37:25.535] DEBUG[15901] chan_sip.c: update_call_counter(5555) - decrement call limit counter on hangup [2009-11-13 15:37:25.535] DEBUG[15901] chan_sip.c: Updating call counter for incoming call [2009-11-13 15:37:25.535] DEBUG[15901] chan_sip.c: Call from peer 'siptrunk1' removed from call limit 1000 [2009-11-13 15:37:25.536] DEBUG[15867] devicestate.c: No provider found, checking channel drivers for SIP - siptrunk1 [2009-11-13 15:37:25.536] DEBUG[15867] chan_sip.c: Checking device state for peer siptrunk1 [2009-11-13 15:37:25.536] DEBUG[15867] devicestate.c: Changing state for SIP/siptrunk1 - state 1 (Not in use) [2009-11-13 15:37:25.536] DEBUG[15867] devicestate.c: device 'SIP/siptrunk1' state '1' [2009-11-13 15:37:25.536] DEBUG[15867] devicestate.c: No provider found, checking channel drivers for SIP - siptrunk1 [2009-11-13 15:37:25.536] DEBUG[15867] chan_sip.c: Checking device state for peer siptrunk1 [2009-11-13 15:37:25.537] DEBUG[15867] devicestate.c: Changing state for SIP/siptrunk1 - state 1 (Not in use) [2009-11-13 15:37:25.537] DEBUG[15867] devicestate.c: device 'SIP/siptrunk1' state '1' [2009-11-13 15:37:25.537] DEBUG[15875] app_queue.c: Device 'SIP/siptrunk1' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [2009-11-13 15:37:25.537] DEBUG[15875] app_queue.c: Device 'SIP/siptrunk1' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [2009-11-13 15:37:26.534] DEBUG[15878] chan_sip.c: Destroying SIP dialog 579116414152@10.0.1.10 [2009-11-13 15:37:26.534] VERBOSE[15878] chan_sip.c: Really destroying SIP dialog '579116414152@10.0.1.10' Method: BYE [2009-11-13 15:38:20.631] VERBOSE[15898] asterisk.c: Beginning asterisk shutdown.... [2009-11-13 15:38:20.631] VERBOSE[15898] asterisk.c: Executing last minute cleanups [2009-11-13 15:38:20.631] VERBOSE[15898] res_musiconhold.c: == Destroying musiconhold processes [2009-11-13 15:38:20.631] DEBUG[15898] res_musiconhold.c: Destroying MOH class 'default' [2009-11-13 15:38:20.631] VERBOSE[15898] asterisk.c: Asterisk cleanly ending (0). [2009-11-13 15:38:20.631] DEBUG[15898] asterisk.c: Asterisk ending (0).