[Jan 25 16:03:00] VERBOSE[5956] logger.c: <--- SIP read from 66.114.80.25:52403 ---> INVITE sip:6469646701@pbxtest.acepbx.com:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.208:5060;branch=z9hG4bK1012C2 From: ;tag=5B4AEC8-118A To: Date: Tue, 25 Jan 2011 21:03:00 GMT Call-ID: 537D0811-27FD11E0-8816CD5D-9D08DBFA@192.168.0.208 Supported: 100rel,timer,replaces Min-SE: 1800 Cisco-Guid: 1342662883-670896608-2282999133-2634603514 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER CSeq: 101 INVITE Max-Forwards: 10 Remote-Party-ID: ;party=calling;screen=no;privacy=off Timestamp: 1295989380 Contact: Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Length: 284 v=0 o=CiscoSystemsSIP-GW-UserAgent 178 2855 IN IP4 192.168.0.208 s=SIP Call c=IN IP4 192.168.0.208 t=0 0 m=audio 16488 RTP/AVP 0 18 101 c=IN IP4 192.168.0.208 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 <-------------> [Jan 25 16:03:00] VERBOSE[5956] logger.c: --- (20 headers 12 lines) --- [Jan 25 16:03:00] DEBUG[5956] chan_sip.c: Setting NAT on RTP to On [Jan 25 16:03:00] DEBUG[5956] chan_sip.c: Setting NAT on UDPTL to On [Jan 25 16:03:00] DEBUG[5956] chan_sip.c: Allocating new SIP dialog for 537D0811-27FD11E0-8816CD5D-9D08DBFA@192.168.0.208 - INVITE (With RTP) [Jan 25 16:03:00] VERBOSE[5956] logger.c: Sending to 66.114.80.25 : 52403 (NAT) [Jan 25 16:03:00] VERBOSE[5956] logger.c: Using INVITE request as basis request - 537D0811-27FD11E0-8816CD5D-9D08DBFA@192.168.0.208 [Jan 25 16:03:00] DEBUG[5956] chan_sip.c: Setting NAT on RTP to On [Jan 25 16:03:00] DEBUG[5956] chan_sip.c: Setting NAT on UDPTL to On [Jan 25 16:03:00] VERBOSE[5956] logger.c: <--- Reliably Transmitting (NAT) to 66.114.80.25:52403 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.0.208:5060;branch=z9hG4bK1012C2;received=66.114.80.25 From: ;tag=5B4AEC8-118A To: ;tag=as6dc1a994 Call-ID: 537D0811-27FD11E0-8816CD5D-9D08DBFA@192.168.0.208 CSeq: 101 INVITE User-Agent: AcePBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="acepbx.com", nonce="19c76cad" Content-Length: 0 <------------> [Jan 25 16:03:00] VERBOSE[5956] logger.c: Scheduling destruction of SIP dialog '537D0811-27FD11E0-8816CD5D-9D08DBFA@192.168.0.208' in 32000 ms (Method: INVITE) [Jan 25 16:03:00] VERBOSE[5956] logger.c: Found user '000011206003' [Jan 25 16:03:00] VERBOSE[5956] logger.c: <--- SIP read from 66.114.80.25:1417 ---> ACK sip:6469646701@pbxtest.acepbx.com:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.208:5060;branch=z9hG4bK1012C2 From: ;tag=5B4AEC8-118A To: ;tag=as6dc1a994 Date: Tue, 25 Jan 2011 21:03:00 GMT Call-ID: 537D0811-27FD11E0-8816CD5D-9D08DBFA@192.168.0.208 Max-Forwards: 10 CSeq: 101 ACK Content-Length: 0 <-------------> [Jan 25 16:03:00] VERBOSE[5956] logger.c: --- (9 headers 0 lines) --- [Jan 25 16:03:00] DEBUG[5956] chan_sip.c: Stopping retransmission on '537D0811-27FD11E0-8816CD5D-9D08DBFA@192.168.0.208' of Response 101: Match Found [Jan 25 16:03:00] VERBOSE[5956] logger.c: <--- SIP read from 66.114.80.25:52403 ---> INVITE sip:6469646701@pbxtest.acepbx.com:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.208:5060;branch=z9hG4bK10131BC2 From: ;tag=5B4AEC8-118A To: Date: Tue, 25 Jan 2011 21:03:00 GMT Call-ID: 537D0811-27FD11E0-8816CD5D-9D08DBFA@192.168.0.208 Supported: 100rel,timer,replaces Min-SE: 1800 Cisco-Guid: 1342662883-670896608-2282999133-2634603514 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER CSeq: 102 INVITE Max-Forwards: 10 Remote-Party-ID: ;party=calling;screen=no;privacy=off Timestamp: 1295989380 Contact: Expires: 180 Allow-Events: telephone-event Proxy-Authorization: Digest username="000011206003",realm="acepbx.com",uri="sip:6469646701@pbxtest.acepbx.com:5060",response="a8c2dfa8d93e89c3bbbc1f7c2157c59d",nonce="19c76cad",algorithm=MD5 Content-Type: application/sdp Content-Length: 284 v=0 o=CiscoSystemsSIP-GW-UserAgent 178 2855 IN IP4 192.168.0.208 s=SIP Call c=IN IP4 192.168.0.208 t=0 0 m=audio 16488 RTP/AVP 0 18 101 c=IN IP4 192.168.0.208 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 <-------------> [Jan 25 16:03:00] VERBOSE[5956] logger.c: --- (21 headers 12 lines) --- [Jan 25 16:03:00] VERBOSE[5956] logger.c: Sending to 66.114.80.25 : 52403 (NAT) [Jan 25 16:03:00] VERBOSE[5956] logger.c: Using INVITE request as basis request - 537D0811-27FD11E0-8816CD5D-9D08DBFA@192.168.0.208 [Jan 25 16:03:00] DEBUG[5956] chan_sip.c: Setting NAT on RTP to On [Jan 25 16:03:00] DEBUG[5956] chan_sip.c: Setting NAT on UDPTL to On [Jan 25 16:03:00] VERBOSE[5956] logger.c: Found user '000011206003' [Jan 25 16:03:00] VERBOSE[5956] logger.c: Found RTP audio format 0 [Jan 25 16:03:00] VERBOSE[5956] logger.c: Found RTP audio format 18 [Jan 25 16:03:00] VERBOSE[5956] logger.c: Found RTP audio format 101 [Jan 25 16:03:00] VERBOSE[5956] logger.c: Found audio description format PCMU for ID 0 [Jan 25 16:03:00] VERBOSE[5956] logger.c: Found audio description format G729 for ID 18 [Jan 25 16:03:00] VERBOSE[5956] logger.c: Found audio description format telephone-event for ID 101 [Jan 25 16:03:00] VERBOSE[5956] logger.c: Capabilities: us - 0x4 (ulaw), peer - audio=0x104 (ulaw|g729)/video=0x0 (nothing), combined - 0x4 (ulaw) [Jan 25 16:03:00] VERBOSE[5956] logger.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Jan 25 16:03:00] DEBUG[5956] chan_sip.c: Our T38 capability = (16160), peer T38 capability (0), joint T38 capability (16160) [Jan 25 16:03:00] VERBOSE[5956] logger.c: Peer audio RTP is at port 192.168.0.208:16488 [Jan 25 16:03:00] DEBUG[5956] chan_sip.c: Peer doesn't provide T.38 UDPTL [Jan 25 16:03:00] DEBUG[5956] chan_sip.c: Checking SIP call limits for device 000011206003 [Jan 25 16:03:00] DEBUG[5956] chan_sip.c: Call from peer '000011206003' is 1 out of 3 [Jan 25 16:03:00] VERBOSE[5956] logger.c: Looking for 6469646701 in xyz (domain pbxtest.acepbx.com) [Jan 25 16:03:00] VERBOSE[5956] logger.c: list_route: hop: [Jan 25 16:03:00] VERBOSE[5956] logger.c: <--- Transmitting (NAT) to 66.114.80.25:52403 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.208:5060;branch=z9hG4bK10131BC2;received=66.114.80.25 From: ;tag=5B4AEC8-118A To: Call-ID: 537D0811-27FD11E0-8816CD5D-9D08DBFA@192.168.0.208 CSeq: 102 INVITE User-Agent: AcePBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: Content-Length: 0 <------------> [Jan 25 16:03:00] DEBUG[8237] pbx.c: Launching 'Set' [Jan 25 16:03:00] VERBOSE[8237] logger.c: -- Executing [6469646701@xyz:1] Set("SIP/000011206003-0000002a", "TIMEOUT(absolute)=10800") in new stack [Jan 25 16:03:00] VERBOSE[8237] logger.c: -- Channel will hangup at 2011-01-26 00:03:00 UTC. [Jan 25 16:03:00] DEBUG[8237] pbx.c: Launching 'Gosub' [Jan 25 16:03:00] VERBOSE[8237] logger.c: -- Executing [6469646701@xyz:2] Gosub("SIP/000011206003-0000002a", "_set_root_channel_id|s|1") in new stack [Jan 25 16:03:00] DEBUG[8237] pbx.c: Function result is '1' [Jan 25 16:03:00] DEBUG[8237] pbx.c: Expression result is '1' [Jan 25 16:03:00] DEBUG[8237] pbx.c: Launching 'GotoIf' [Jan 25 16:03:00] VERBOSE[8237] logger.c: -- Executing [s@_set_root_channel_id:1] GotoIf("SIP/000011206003-0000002a", "1?+1:done") in new stack [Jan 25 16:03:00] VERBOSE[8237] logger.c: -- Goto (_set_root_channel_id,s,2) [Jan 25 16:03:00] DEBUG[8237] pbx.c: Launching 'Set' [Jan 25 16:03:00] VERBOSE[8237] logger.c: -- Executing [s@_set_root_channel_id:2] Set("SIP/000011206003-0000002a", "__FIRST_CHANNEL_ID=1295989380.42") in new stack [Jan 25 16:03:00] DEBUG[8237] pbx.c: Launching 'Set' [Jan 25 16:03:00] VERBOSE[8237] logger.c: -- Executing [s@_set_root_channel_id:3] Set("SIP/000011206003-0000002a", "__FIRST_CHANNEL_NAME=SIP/000011206003-0000002a") in new stack [Jan 25 16:03:00] DEBUG[8237] pbx.c: Launching 'Return' [Jan 25 16:03:00] VERBOSE[8237] logger.c: -- Executing [s@_set_root_channel_id:4] Return("SIP/000011206003-0000002a", "") in new stack [Jan 25 16:03:00] DEBUG[8237] pbx.c: Launching 'AGI' [Jan 25 16:03:00] VERBOSE[8237] logger.c: -- Executing [6469646701@xyz:3] AGI("SIP/000011206003-0000002a", "agi://pbxtest/CP?blnd_xfer=") in new stack [Jan 25 16:03:00] DEBUG[8237] pbx.c: Function result is '000011206003' [Jan 25 16:03:00] DEBUG[8237] pbx.c: Function result is '0' [Jan 25 16:03:00] DEBUG[8237] pbx.c: Function result is '0' [Jan 25 16:03:00] DEBUG[8237] pbx.c: Function result is '0' [Jan 25 16:03:00] DEBUG[8237] pbx.c: Function result is '0' [Jan 25 16:03:00] VERBOSE[8237] logger.c: -- AGI Script Executing Application: (Set) Options: (GROUP()=EXT_LINE@LINES_COMPANY_2_OUT) [Jan 25 16:03:00] VERBOSE[8237] logger.c: -- AGI Script Executing Application: (SetCallerPres) Options: (allowed_passed_screen) [Jan 25 16:03:00] VERBOSE[8237] logger.c: -- AGI Script Executing Application: (Set) Options: (CALLERID(name)=7189282005) [Jan 25 16:03:00] VERBOSE[8237] logger.c: -- AGI Script Executing Application: (Set) Options: (CALLERID(num)=7189282005) [Jan 25 16:03:00] VERBOSE[8237] logger.c: -- AGI Script Executing Application: (Dial) Options: (SIP/6469646701@sipp1|180|M(call-connect-external)) [Jan 25 16:03:00] DEBUG[8237] chan_sip.c: Asked to create a SIP channel with formats: 0x4 (ulaw) [Jan 25 16:03:00] DEBUG[8237] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) [Jan 25 16:03:00] DEBUG[8237] chan_sip.c: Setting NAT on RTP to On [Jan 25 16:03:00] DEBUG[8237] chan_sip.c: Setting NAT on UDPTL to On [Jan 25 16:03:00] DEBUG[8237] channel.c: Not copying variable DIALEDTIME. [Jan 25 16:03:00] DEBUG[8237] channel.c: Not copying variable ANSWEREDTIME. [Jan 25 16:03:00] DEBUG[8237] channel.c: Not copying variable DIALEDPEERNAME. [Jan 25 16:03:00] DEBUG[8237] channel.c: Not copying variable DIALEDPEERNUMBER. [Jan 25 16:03:00] DEBUG[8237] channel.c: Not copying variable DIALSTATUS. [Jan 25 16:03:00] DEBUG[8237] channel.c: Copying hard-transferable variable DYNAMIC_FEATURES. [Jan 25 16:03:00] DEBUG[8237] channel.c: Copying soft-transferable variable UF. [Jan 25 16:03:00] DEBUG[8237] channel.c: Not copying variable LINE_COMPANY_SEIZED_OUT. [Jan 25 16:03:00] DEBUG[8237] channel.c: Copying hard-transferable variable FIRST_CHANNEL_NAME. [Jan 25 16:03:00] DEBUG[8237] channel.c: Copying hard-transferable variable FIRST_CHANNEL_ID. [Jan 25 16:03:00] DEBUG[8237] channel.c: Not copying variable SIPCALLID. [Jan 25 16:03:00] DEBUG[8237] channel.c: Not copying variable SIPUSERAGENT. [Jan 25 16:03:00] DEBUG[8237] channel.c: Not copying variable SIPDOMAIN. [Jan 25 16:03:00] DEBUG[8237] channel.c: Not copying variable SIPURI. [Jan 25 16:03:00] DEBUG[8237] chan_sip.c: Outgoing Call for 6469646701 [Jan 25 16:03:00] VERBOSE[8237] logger.c: Audio is at 66.114.80.26 port 48358 [Jan 25 16:03:00] VERBOSE[8237] logger.c: Adding codec 0x4 (ulaw) to SDP [Jan 25 16:03:00] VERBOSE[8237] logger.c: Adding non-codec 0x1 (telephone-event) to SDP [Jan 25 16:03:00] VERBOSE[8237] logger.c: Reliably Transmitting (NAT) to 66.114.83.7:5060: INVITE sip:6469646701@66.114.83.7 SIP/2.0 Via: SIP/2.0/UDP 66.114.80.26:5060;branch=z9hG4bK707e7b2d;rport From: "7189282005" ;tag=as57f27f73 To: Contact: Call-ID: 7d7cd6e9419a8b4217b6142c1631edeb@66.114.80.26 CSeq: 102 INVITE User-Agent: AcePBX Max-Forwards: 70 Remote-Party-ID: "7189282005" ;privacy=off;screen=yes Date: Tue, 25 Jan 2011 21:03:00 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Type: application/sdp Content-Length: 211 v=0 o=root 5946 5946 IN IP4 66.114.80.26 s=session c=IN IP4 66.114.80.26 t=0 0 m=audio 48358 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Jan 25 16:03:00] VERBOSE[8237] logger.c: -- Called 6469646701@sipp1 [Jan 25 16:03:00] VERBOSE[5956] logger.c: <--- SIP read from 66.114.83.7:5060 ---> SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 66.114.80.26:5060;branch=z9hG4bK707e7b2d;rport=5060 From: "7189282005" ;tag=as57f27f73 To: Call-ID: 7d7cd6e9419a8b4217b6142c1631edeb@66.114.80.26 CSeq: 102 INVITE Content-Length: 0 <-------------> [Jan 25 16:03:00] VERBOSE[5956] logger.c: --- (7 headers 0 lines) --- [Jan 25 16:03:00] DEBUG[5956] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '7d7cd6e9419a8b4217b6142c1631edeb@66.114.80.26' Request 102: Found [Jan 25 16:03:03] VERBOSE[5956] logger.c: <--- SIP read from 66.114.80.25:52403 ---> REGISTER sip:pbxtest.acepbx.com:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.208:5060;branch=z9hG4bK10141A34 From: "247181201" ;tag=5B4B9B8-C98 To: "247181201" Date: Tue, 25 Jan 2011 21:03:03 GMT Call-ID: 89D0F9EC-27FA11E0-87E9CD5D-9D08DBFA User-Agent: Cisco-SIPGateway/IOS-12.x Max-Forwards: 10 Timestamp: 1295989383 CSeq: 11 REGISTER Contact: Expires: 120 Content-Length: 0 <-------------> [Jan 25 16:03:03] VERBOSE[5956] logger.c: --- (13 headers 0 lines) --- [Jan 25 16:03:03] DEBUG[5956] chan_sip.c: Allocating new SIP dialog for 89D0F9EC-27FA11E0-87E9CD5D-9D08DBFA - REGISTER (No RTP) [Jan 25 16:03:03] VERBOSE[5956] logger.c: Using latest REGISTER request as basis request [Jan 25 16:03:03] VERBOSE[5956] logger.c: Sending to 66.114.80.25 : 52403 (NAT) [Jan 25 16:03:03] VERBOSE[5956] logger.c: <--- Transmitting (NAT) to 66.114.80.25:52403 ---> SIP/2.0 404 Not found Via: SIP/2.0/UDP 192.168.0.208:5060;branch=z9hG4bK10141A34;received=66.114.80.25 From: "247181201" ;tag=5B4B9B8-C98 To: "247181201" ;tag=as1c76d6a4 Call-ID: 89D0F9EC-27FA11E0-87E9CD5D-9D08DBFA CSeq: 11 REGISTER User-Agent: AcePBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 <------------> [Jan 25 16:03:03] DEBUG[5956] chan_sip.c: Registration from '"247181201" ' failed for '66.114.80.25' - No matching peer found [Jan 25 16:03:03] VERBOSE[5956] logger.c: Scheduling destruction of SIP dialog '89D0F9EC-27FA11E0-87E9CD5D-9D08DBFA' in 32000 ms (Method: REGISTER) [Jan 25 16:03:04] NOTICE[5956] chan_sip.c: -- Re-registration for pbxtest@66.114.83.7 [Jan 25 16:03:04] DEBUG[5956] chan_sip.c: Allocating new SIP dialog for 70d42e23493ff52b3b19df1921b16433@66.114.80.26 - REGISTER (No RTP) [Jan 25 16:03:04] DEBUG[5956] chan_sip.c: Scheduled a registration timeout for 66.114.83.7 id #46815 [Jan 25 16:03:04] DEBUG[5956] chan_sip.c: >>> Re-using Auth data for pbxtest@66.114.83.7 [Jan 25 16:03:04] VERBOSE[5956] logger.c: REGISTER 12 headers, 0 lines [Jan 25 16:03:04] VERBOSE[5956] logger.c: Reliably Transmitting (NAT) to 66.114.83.7:5060: REGISTER sip:66.114.83.7 SIP/2.0 Via: SIP/2.0/UDP 66.114.80.26:5060;branch=z9hG4bK3c199daa;rport From: ;tag=as677ca258 To: Call-ID: 70d42e23493ff52b3b19df1921b16433@66.114.80.26 CSeq: 231 REGISTER User-Agent: AcePBX Max-Forwards: 70 Authorization: Digest username="pbxtest", realm="acepbx.com", algorithm=MD5, uri="sip:66.114.83.7", nonce="4d3f3a79f0f038437d15fa475f23de1dc330025b", response="f4f3f82fc9045693aee7f478d25fd138", qop=auth, cnonce="2d4c11d4", nc=00000004 Expires: 120 Contact: Event: registration Content-Length: 0 --- [Jan 25 16:03:04] VERBOSE[5956] logger.c: <--- SIP read from 66.114.83.7:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 66.114.80.26:5060;branch=z9hG4bK3c199daa;rport=5060 From: ;tag=as677ca258 To: Call-ID: 70d42e23493ff52b3b19df1921b16433@66.114.80.26 CSeq: 231 REGISTER Content-Length: 0 <-------------> [Jan 25 16:03:04] VERBOSE[5956] logger.c: --- (7 headers 0 lines) --- [Jan 25 16:03:04] DEBUG[5956] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '70d42e23493ff52b3b19df1921b16433@66.114.80.26' Request 231: Found [Jan 25 16:03:04] VERBOSE[5956] logger.c: <--- SIP read from 66.114.83.7:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 66.114.80.26:5060;branch=z9hG4bK3c199daa;rport=5060 From: ;tag=as677ca258 To: ;tag=bb62d7e482cc1e58bd69683900d56829.c777 Call-ID: 70d42e23493ff52b3b19df1921b16433@66.114.80.26 CSeq: 231 REGISTER WWW-Authenticate: Digest realm="acepbx.com", nonce="4d3f3bb49142ab783ed20f0b540e4857521fea8c", qop="auth", stale=true Content-Length: 0 <-------------> [Jan 25 16:03:04] VERBOSE[5956] logger.c: --- (8 headers 0 lines) --- [Jan 25 16:03:04] DEBUG[5956] chan_sip.c: Stopping retransmission on '70d42e23493ff52b3b19df1921b16433@66.114.80.26' of Request 231: Match Found [Jan 25 16:03:04] VERBOSE[5956] logger.c: Responding to challenge, registration to domain/host name 66.114.83.7 [Jan 25 16:03:04] DEBUG[5956] chan_sip.c: Initializing already initialized SIP dialog 70d42e23493ff52b3b19df1921b16433@66.114.80.26 (presumably reinvite) [Jan 25 16:03:04] VERBOSE[5956] logger.c: REGISTER 12 headers, 0 lines [Jan 25 16:03:04] VERBOSE[5956] logger.c: Reliably Transmitting (NAT) to 66.114.83.7:5060: REGISTER sip:66.114.83.7 SIP/2.0 Via: SIP/2.0/UDP 66.114.80.26:5060;branch=z9hG4bK17b0135b;rport From: ;tag=as05a17454 To: Call-ID: 70d42e23493ff52b3b19df1921b16433@66.114.80.26 CSeq: 232 REGISTER User-Agent: AcePBX Max-Forwards: 70 Authorization: Digest username="pbxtest", realm="acepbx.com", algorithm=MD5, uri="sip:66.114.83.7", nonce="4d3f3bb49142ab783ed20f0b540e4857521fea8c", response="2fa4afe722f1c8bba2ff34bc16266ab7", qop=auth, cnonce="02666f12", nc=00000001 Expires: 120 Contact: Event: registration Content-Length: 0 --- [Jan 25 16:03:04] VERBOSE[5956] logger.c: <--- SIP read from 66.114.83.7:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 66.114.80.26:5060;branch=z9hG4bK17b0135b;rport=5060 From: ;tag=as05a17454 To: Call-ID: 70d42e23493ff52b3b19df1921b16433@66.114.80.26 CSeq: 232 REGISTER Content-Length: 0 <-------------> [Jan 25 16:03:04] VERBOSE[5956] logger.c: --- (7 headers 0 lines) --- [Jan 25 16:03:04] DEBUG[5956] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '70d42e23493ff52b3b19df1921b16433@66.114.80.26' Request 232: Found [Jan 25 16:03:04] VERBOSE[5956] logger.c: <--- SIP read from 66.114.83.7:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 66.114.80.26:5060;branch=z9hG4bK17b0135b;rport=5060 From: ;tag=as05a17454 To: ;tag=bb62d7e482cc1e58bd69683900d56829.0e01 Call-ID: 70d42e23493ff52b3b19df1921b16433@66.114.80.26 CSeq: 232 REGISTER Contact: ;expires=120 Content-Length: 0 <-------------> [Jan 25 16:03:04] VERBOSE[5956] logger.c: --- (8 headers 0 lines) --- [Jan 25 16:03:04] DEBUG[5956] chan_sip.c: Stopping retransmission on '70d42e23493ff52b3b19df1921b16433@66.114.80.26' of Request 232: Match Found [Jan 25 16:03:04] DEBUG[5956] chan_sip.c: Registration successful [Jan 25 16:03:04] DEBUG[5956] chan_sip.c: Cancelling timeout 46815 [Jan 25 16:03:04] VERBOSE[5956] logger.c: Scheduling destruction of SIP dialog '70d42e23493ff52b3b19df1921b16433@66.114.80.26' in 32000 ms (Method: REGISTER) [Jan 25 16:03:04] NOTICE[5956] chan_sip.c: Outbound Registration: Expiry for 66.114.83.7 is 120 sec (Scheduling reregistration in 105 s) [Jan 25 16:03:14] VERBOSE[5956] logger.c: <--- SIP read from 66.114.83.7:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 66.114.80.26:5060;branch=z9hG4bK707e7b2d;rport=5060 From: "7189282005" ;tag=as57f27f73 To: ;tag=7CD92148-202C Date: Tue, 25 Jan 2011 21:03:00 GMT Call-ID: 7d7cd6e9419a8b4217b6142c1631edeb@66.114.80.26 Server: Cisco-SIPGateway/IOS-12.x CSeq: 102 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER Allow-Events: telephone-event Contact: Record-Route: Content-Disposition: session;handling=required Content-Type: application/sdp Content-Length: 250 v=0 o=CiscoSystemsSIP-GW-UserAgent 5748 1015 IN IP4 66.114.76.201 s=SIP Call c=IN IP4 66.114.76.201 t=0 0 m=audio 19434 RTP/AVP 0 101 c=IN IP4 66.114.76.201 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 <-------------> [Jan 25 16:03:14] VERBOSE[5956] logger.c: --- (15 headers 11 lines) --- [Jan 25 16:03:14] DEBUG[5956] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '7d7cd6e9419a8b4217b6142c1631edeb@66.114.80.26' Request 102: Found [Jan 25 16:03:14] VERBOSE[5956] logger.c: Found RTP audio format 0 [Jan 25 16:03:14] VERBOSE[5956] logger.c: Found RTP audio format 101 [Jan 25 16:03:14] VERBOSE[5956] logger.c: Found audio description format PCMU for ID 0 [Jan 25 16:03:14] VERBOSE[5956] logger.c: Found audio description format telephone-event for ID 101 [Jan 25 16:03:14] VERBOSE[5956] logger.c: Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) [Jan 25 16:03:14] VERBOSE[5956] logger.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Jan 25 16:03:14] DEBUG[5956] chan_sip.c: Our T38 capability = (16160), peer T38 capability (0), joint T38 capability (16160) [Jan 25 16:03:14] VERBOSE[5956] logger.c: Peer audio RTP is at port 66.114.76.201:19434 [Jan 25 16:03:14] DEBUG[5956] chan_sip.c: Peer doesn't provide T.38 UDPTL [Jan 25 16:03:14] VERBOSE[8237] logger.c: -- SIP/sipp1-0000002b is making progress passing it to SIP/000011206003-0000002a [Jan 25 16:03:14] DEBUG[8237] chan_sip.c: Setting framing from config on incoming call [Jan 25 16:03:14] VERBOSE[8237] logger.c: Audio is at 66.114.80.26 port 45294 [Jan 25 16:03:14] VERBOSE[8237] logger.c: Adding codec 0x4 (ulaw) to SDP [Jan 25 16:03:14] VERBOSE[8237] logger.c: Adding non-codec 0x1 (telephone-event) to SDP [Jan 25 16:03:14] VERBOSE[8237] logger.c: <--- Transmitting (NAT) to 66.114.80.25:52403 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.0.208:5060;branch=z9hG4bK10131BC2;received=66.114.80.25 From: ;tag=5B4AEC8-118A To: ;tag=as2e186f49 Call-ID: 537D0811-27FD11E0-8816CD5D-9D08DBFA@192.168.0.208 CSeq: 102 INVITE User-Agent: AcePBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: Content-Type: application/sdp Content-Length: 211 v=0 o=root 5946 5946 IN IP4 66.114.80.26 s=session c=IN IP4 66.114.80.26 t=0 0 m=audio 45294 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <------------> [Jan 25 16:03:14] DEBUG[8237] rtp.c: RTP NAT: Got audio from other end. Now sending to address 66.114.80.25:16488 [Jan 25 16:03:14] DEBUG[8237] rtp.c: Ooh, format changed from unknown to ulaw [Jan 25 16:03:14] DEBUG[8237] rtp.c: Created smoother: format: 4 ms: 20 len: 160 [Jan 25 16:03:14] DEBUG[8237] rtp.c: Ooh, format changed from unknown to ulaw [Jan 25 16:03:14] DEBUG[8237] rtp.c: Created smoother: format: 4 ms: 20 len: 160 [Jan 25 16:03:17] DEBUG[8237] rtp.c: Got RTCP report of 132 bytes [Jan 25 16:03:17] DEBUG[8237] rtp.c: Got RTCP report of 132 bytes [Jan 25 16:03:18] VERBOSE[5956] logger.c: <--- SIP read from 66.114.83.7:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 66.114.80.26:5060;branch=z9hG4bK707e7b2d;rport=5060 From: "7189282005" ;tag=as57f27f73 To: ;tag=7CD92148-202C Date: Tue, 25 Jan 2011 21:03:00 GMT Call-ID: 7d7cd6e9419a8b4217b6142c1631edeb@66.114.80.26 Server: Cisco-SIPGateway/IOS-12.x CSeq: 102 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER Supported: replaces Allow-Events: telephone-event Contact: Record-Route: P-AcePBX-TermGW: alias=vgw1 Content-Type: application/sdp Content-Length: 250 v=0 o=CiscoSystemsSIP-GW-UserAgent 5748 1015 IN IP4 66.114.76.201 s=SIP Call c=IN IP4 66.114.76.201 t=0 0 m=audio 19434 RTP/AVP 0 101 c=IN IP4 66.114.76.201 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 <-------------> [Jan 25 16:03:18] VERBOSE[5956] logger.c: --- (16 headers 11 lines) --- [Jan 25 16:03:18] DEBUG[5956] chan_sip.c: Acked pending invite 102 [Jan 25 16:03:18] DEBUG[5956] chan_sip.c: Stopping retransmission on '7d7cd6e9419a8b4217b6142c1631edeb@66.114.80.26' of Request 102: Match Found [Jan 25 16:03:18] VERBOSE[5956] logger.c: Found RTP audio format 0 [Jan 25 16:03:18] VERBOSE[5956] logger.c: Found RTP audio format 101 [Jan 25 16:03:18] VERBOSE[5956] logger.c: Found audio description format PCMU for ID 0 [Jan 25 16:03:18] VERBOSE[5956] logger.c: Found audio description format telephone-event for ID 101 [Jan 25 16:03:18] VERBOSE[5956] logger.c: Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) [Jan 25 16:03:18] VERBOSE[5956] logger.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Jan 25 16:03:18] DEBUG[5956] chan_sip.c: Our T38 capability = (16160), peer T38 capability (0), joint T38 capability (16160) [Jan 25 16:03:18] VERBOSE[5956] logger.c: Peer audio RTP is at port 66.114.76.201:19434 [Jan 25 16:03:18] DEBUG[5956] chan_sip.c: Peer doesn't provide T.38 UDPTL [Jan 25 16:03:18] VERBOSE[5956] logger.c: list_route: hop: [Jan 25 16:03:18] VERBOSE[5956] logger.c: set_destination: Parsing for address/port to send to [Jan 25 16:03:18] VERBOSE[5956] logger.c: set_destination: set destination to 66.114.83.7, port 5060 [Jan 25 16:03:18] VERBOSE[5956] logger.c: Transmitting (NAT) to 66.114.83.7:5060: ACK sip:000016469646701@66.114.76.201:5060 SIP/2.0 Via: SIP/2.0/UDP 66.114.80.26:5060;branch=z9hG4bK3cd36aab;rport Route: From: "7189282005" ;tag=as57f27f73 To: ;tag=7CD92148-202C Contact: Call-ID: 7d7cd6e9419a8b4217b6142c1631edeb@66.114.80.26 CSeq: 102 ACK User-Agent: AcePBX Max-Forwards: 70 Remote-Party-ID: "7189282005" ;privacy=off;screen=yes Content-Length: 0 --- [Jan 25 16:03:18] VERBOSE[8237] logger.c: -- SIP/sipp1-0000002b answered SIP/000011206003-0000002a [Jan 25 16:03:18] DEBUG[8237] pbx.c: Function result is 'sip:000016469646701@66.114.76.201:5060' [Jan 25 16:03:18] DEBUG[8237] pbx.c: Launching 'Set' [Jan 25 16:03:18] VERBOSE[8237] logger.c: -- Executing [s@macro-call-connect-external:1] Set("SIP/sipp1-0000002b", "EXT_CONTACT_URI=sip:000016469646701@66.114.76.201:5060") in new stack [Jan 25 16:03:18] DEBUG[8237] app_macro.c: Executed application: Set [Jan 25 16:03:18] DEBUG[8237] app_dial.c: Macro exited with status 0 [Jan 25 16:03:18] DEBUG[8237] chan_sip.c: SIP answering channel: SIP/000011206003-0000002a [Jan 25 16:03:18] DEBUG[8237] chan_sip.c: Setting framing from config on incoming call [Jan 25 16:03:18] VERBOSE[8237] logger.c: Audio is at 66.114.80.26 port 45294 [Jan 25 16:03:18] VERBOSE[8237] logger.c: Adding codec 0x4 (ulaw) to SDP [Jan 25 16:03:18] VERBOSE[8237] logger.c: Adding non-codec 0x1 (telephone-event) to SDP [Jan 25 16:03:18] VERBOSE[8237] logger.c: <--- Reliably Transmitting (NAT) to 66.114.80.25:52403 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.208:5060;branch=z9hG4bK10131BC2;received=66.114.80.25 From: ;tag=5B4AEC8-118A To: ;tag=as2e186f49 Call-ID: 537D0811-27FD11E0-8816CD5D-9D08DBFA@192.168.0.208 CSeq: 102 INVITE User-Agent: AcePBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: Content-Type: application/sdp Content-Length: 211 v=0 o=root 5946 5947 IN IP4 66.114.80.26 s=session c=IN IP4 66.114.80.26 t=0 0 m=audio 45294 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <------------> [Jan 25 16:03:19] VERBOSE[5956] logger.c: <--- SIP read from 66.114.80.25:52403 ---> ACK sip:6469646701@66.114.80.26:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.208:5060;branch=z9hG4bK10158E From: ;tag=5B4AEC8-118A To: ;tag=as2e186f49 Date: Tue, 25 Jan 2011 21:03:00 GMT Call-ID: 537D0811-27FD11E0-8816CD5D-9D08DBFA@192.168.0.208 Max-Forwards: 10 CSeq: 102 ACK Proxy-Authorization: Digest username="000011206003",realm="acepbx.com",uri="sip:6469646701@pbxtest.acepbx.com:5060",response="a8c2dfa8d93e89c3bbbc1f7c2157c59d",nonce="19c76cad",algorithm=MD5 Content-Length: 0 <-------------> [Jan 25 16:03:19] VERBOSE[5956] logger.c: --- (10 headers 0 lines) --- [Jan 25 16:03:19] DEBUG[5956] chan_sip.c: Stopping retransmission on '537D0811-27FD11E0-8816CD5D-9D08DBFA@192.168.0.208' of Response 102: Match Found [Jan 25 16:03:19] DEBUG[8237] rtp.c: Got RTCP report of 132 bytes [Jan 25 16:03:19] DEBUG[8237] rtp.c: Got RTCP report of 132 bytes [Jan 25 16:03:21] VERBOSE[5956] logger.c: <--- SIP read from 66.114.83.7:5060 ---> INVITE sip:7189282005@66.114.80.26:5060 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 66.114.83.7;branch=z9hG4bK3786.05b76082.0 Via: SIP/2.0/UDP 66.114.76.201:5060;branch=z9hG4bK1639511725 From: ;tag=7CD92148-202C To: "7189282005" ;tag=as57f27f73 Date: Tue, 25 Jan 2011 21:03:21 GMT Call-ID: 7d7cd6e9419a8b4217b6142c1631edeb@66.114.80.26 Supported: 100rel,timer,replaces Min-SE: 1800 Cisco-Guid: 1402681193-670896608-2180817809-1960822137 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER CSeq: 101 INVITE Max-Forwards: 9 Timestamp: 1295989401 Contact: Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Length: 399 v=0 o=CiscoSystemsSIP-GW-UserAgent 5748 1016 IN IP4 66.114.76.201 s=SIP Call c=IN IP4 66.114.76.201 t=0 0 m=image 19434 udptl t38 c=IN IP4 66.114.76.201 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxFillBitRemoval:0 a=T38FaxTranscodingMMR:0 a=T38FaxTranscodingJBIG:0 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:200 a=T38FaxMaxDatagram:72 a=T38FaxUdpEC:t38UDPRedundancy <-------------> [Jan 25 16:03:21] VERBOSE[5956] logger.c: --- (21 headers 16 lines) --- [Jan 25 16:03:21] VERBOSE[5956] logger.c: Sending to 66.114.83.7 : 5060 (NAT) [Jan 25 16:03:21] VERBOSE[5956] logger.c: Got T.38 offer in SDP in dialog 7d7cd6e9419a8b4217b6142c1631edeb@66.114.80.26 [Jan 25 16:03:21] VERBOSE[5956] logger.c: Capabilities: us - 0x4 (ulaw), peer - audio=0x0 (nothing)/video=0x0 (nothing), combined - 0x0 (nothing) [Jan 25 16:03:21] VERBOSE[5956] logger.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) [Jan 25 16:03:21] DEBUG[5956] chan_sip.c: Our T38 capability = (16160), peer T38 capability (16160), joint T38 capability (16160) [Jan 25 16:03:21] VERBOSE[5956] logger.c: Got T.38 Re-invite without audio. Keeping RTP active during T.38 session. [Jan 25 16:03:21] DEBUG[5956] chan_sip.c: Peer T.38 UDPTL is at port 66.114.76.201:19434 [Jan 25 16:03:21] VERBOSE[5956] logger.c: <--- Transmitting (NAT) to 66.114.83.7:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 66.114.83.7;branch=z9hG4bK3786.05b76082.0;received=66.114.83.7 Via: SIP/2.0/UDP 66.114.76.201:5060;branch=z9hG4bK1639511725 Record-Route: From: ;tag=7CD92148-202C To: "7189282005" ;tag=as57f27f73 Call-ID: 7d7cd6e9419a8b4217b6142c1631edeb@66.114.80.26 CSeq: 101 INVITE User-Agent: AcePBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: Content-Length: 0 <------------> [Jan 25 16:03:21] DEBUG[5956] chan_sip.c: Strict routing enforced for session 537D0811-27FD11E0-8816CD5D-9D08DBFA@192.168.0.208 [Jan 25 16:03:21] VERBOSE[5956] logger.c: set_destination: Parsing for address/port to send to [Jan 25 16:03:21] VERBOSE[5956] logger.c: set_destination: set destination to 192.168.0.208, port 5060 [Jan 25 16:03:21] DEBUG[5956] chan_sip.c: T.38 UDPTL is at 66.114.80.26 port 40579 [Jan 25 16:03:21] DEBUG[5956] chan_sip.c: Our T38 capability (16160), peer T38 capability (16160), joint capability (16160) [Jan 25 16:03:21] DEBUG[5956] chan_sip.c: Initializing already initialized SIP dialog 537D0811-27FD11E0-8816CD5D-9D08DBFA@192.168.0.208 (presumably reinvite) [Jan 25 16:03:21] VERBOSE[5956] logger.c: Reliably Transmitting (NAT) to 66.114.80.25:52403: INVITE sip:000011206003@192.168.0.208:5060 SIP/2.0 Via: SIP/2.0/UDP 66.114.80.26:5060;branch=z9hG4bK4c03f88c;rport From: ;tag=as2e186f49 To: ;tag=5B4AEC8-118A Contact: Call-ID: 537D0811-27FD11E0-8816CD5D-9D08DBFA@192.168.0.208 CSeq: 102 INVITE User-Agent: AcePBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces X-asterisk-info: SIP re-invite (T38 switchover) Content-Type: application/sdp Content-Length: 268 v=0 o=root 5946 5948 IN IP4 66.114.80.26 s=session c=IN IP4 66.114.80.26 t=0 0 m=image 40579 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:72 a=T38FaxMaxDatagram:72 a=T38FaxUdpEC:t38UDPRedundancy --- [Jan 25 16:03:21] VERBOSE[5956] logger.c: <--- SIP read from 66.114.80.25:52403 ---> INVITE sip:6469646701@66.114.80.26:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.208:5060;branch=z9hG4bK10162637 From: ;tag=5B4AEC8-118A To: ;tag=as2e186f49 Date: Tue, 25 Jan 2011 21:03:21 GMT Call-ID: 537D0811-27FD11E0-8816CD5D-9D08DBFA@192.168.0.208 Supported: 100rel,timer,replaces Min-SE: 1800 Cisco-Guid: 1342662883-670896608-2282999133-2634603514 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER CSeq: 103 INVITE Max-Forwards: 10 Remote-Party-ID: ;party=calling;screen=no;privacy=off Timestamp: 1295989401 Contact: Expires: 180 Allow-Events: telephone-event Proxy-Authorization: Digest username="000011206003",realm="acepbx.com",uri="sip:6469646701@66.114.80.26:5060",response="28dd26fefb91468d95b8896ccfe54ca2",nonce="19c76cad",algorithm=MD5 Content-Type: application/sdp Content-Length: 398 v=0 o=CiscoSystemsSIP-GW-UserAgent 178 2856 IN IP4 192.168.0.208 s=SIP Call c=IN IP4 192.168.0.208 t=0 0 m=image 16488 udptl t38 c=IN IP4 192.168.0.208 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxFillBitRemoval:0 a=T38FaxTranscodingMMR:0 a=T38FaxTranscodingJBIG:0 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:200 a=T38FaxMaxDatagram:72 a=T38FaxUdpEC:t38UDPRedundancy <-------------> [Jan 25 16:03:21] VERBOSE[5956] logger.c: --- (21 headers 16 lines) --- [Jan 25 16:03:21] WARNING[5956] chan_sip.c: Failed to read an alternate host or port in SDP. Expect audio problems [Jan 25 16:03:21] WARNING[5956] chan_sip.c: Failed to set an alternate media source on glared reinvite. Audio may not work properly on this call. [Jan 25 16:03:21] VERBOSE[5956] logger.c: <--- Reliably Transmitting (NAT) to 66.114.80.25:52403 ---> SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 192.168.0.208:5060;branch=z9hG4bK10162637;received=66.114.80.25 From: ;tag=5B4AEC8-118A To: ;tag=as2e186f49 Call-ID: 537D0811-27FD11E0-8816CD5D-9D08DBFA@192.168.0.208 CSeq: 103 INVITE User-Agent: AcePBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 <------------> [Jan 25 16:03:21] DEBUG[5956] chan_sip.c: Got INVITE on call where we already have pending INVITE, deferring that - 537D0811-27FD11E0-8816CD5D-9D08DBFA@192.168.0.208 [Jan 25 16:03:21] VERBOSE[5956] logger.c: <--- SIP read from 66.114.80.25:1417 ---> SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 66.114.80.26:5060;branch=z9hG4bK4c03f88c;rport From: ;tag=as2e186f49 To: ;tag=5B4AEC8-118A Call-ID: 537D0811-27FD11E0-8816CD5D-9D08DBFA@192.168.0.208 CSeq: 102 INVITE Content-Length: 0 <-------------> [Jan 25 16:03:21] VERBOSE[5956] logger.c: --- (7 headers 0 lines) --- [Jan 25 16:03:21] DEBUG[5956] chan_sip.c: Acked pending invite 102 [Jan 25 16:03:21] DEBUG[5956] chan_sip.c: Stopping retransmission on '537D0811-27FD11E0-8816CD5D-9D08DBFA@192.168.0.208' of Request 102: Match Found [Jan 25 16:03:21] DEBUG[5956] chan_sip.c: Strict routing enforced for session 537D0811-27FD11E0-8816CD5D-9D08DBFA@192.168.0.208 [Jan 25 16:03:21] VERBOSE[5956] logger.c: set_destination: Parsing for address/port to send to [Jan 25 16:03:21] VERBOSE[5956] logger.c: set_destination: set destination to 192.168.0.208, port 5060 [Jan 25 16:03:21] VERBOSE[5956] logger.c: Transmitting (NAT) to 66.114.80.25:1417: ACK sip:000011206003@192.168.0.208:5060 SIP/2.0 Via: SIP/2.0/UDP 66.114.80.26:5060;branch=z9hG4bK4c03f88c;rport From: ;tag=as2e186f49 To: ;tag=5B4AEC8-118A Contact: Call-ID: 537D0811-27FD11E0-8816CD5D-9D08DBFA@192.168.0.208 CSeq: 102 ACK User-Agent: AcePBX Max-Forwards: 70 Content-Length: 0 --- [Jan 25 16:03:21] VERBOSE[5956] logger.c: <--- SIP read from 66.114.80.25:1417 ---> ACK sip:6469646701@66.114.80.26:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.208:5060;branch=z9hG4bK10162637 From: ;tag=5B4AEC8-118A To: ;tag=as2e186f49 Date: Tue, 25 Jan 2011 21:03:21 GMT Call-ID: 537D0811-27FD11E0-8816CD5D-9D08DBFA@192.168.0.208 Max-Forwards: 10 CSeq: 103 ACK Content-Length: 0 <-------------> [Jan 25 16:03:21] VERBOSE[5956] logger.c: --- (9 headers 0 lines) --- [Jan 25 16:03:21] DEBUG[5956] chan_sip.c: Stopping retransmission on '537D0811-27FD11E0-8816CD5D-9D08DBFA@192.168.0.208' of Response 103: Match Found [Jan 25 16:03:23] DEBUG[5956] chan_sip.c: Sending pending reinvite on '537D0811-27FD11E0-8816CD5D-9D08DBFA@192.168.0.208' [Jan 25 16:03:23] DEBUG[5956] chan_sip.c: Strict routing enforced for session 537D0811-27FD11E0-8816CD5D-9D08DBFA@192.168.0.208 [Jan 25 16:03:23] VERBOSE[5956] logger.c: set_destination: Parsing for address/port to send to [Jan 25 16:03:23] VERBOSE[5956] logger.c: set_destination: set destination to 192.168.0.208, port 5060 [Jan 25 16:03:23] VERBOSE[5956] logger.c: Audio is at 66.114.80.26 port 45294 [Jan 25 16:03:23] VERBOSE[5956] logger.c: Adding codec 0x4 (ulaw) to SDP [Jan 25 16:03:23] VERBOSE[5956] logger.c: Adding non-codec 0x1 (telephone-event) to SDP [Jan 25 16:03:23] DEBUG[5956] chan_sip.c: Initializing already initialized SIP dialog 537D0811-27FD11E0-8816CD5D-9D08DBFA@192.168.0.208 (presumably reinvite) [Jan 25 16:03:23] VERBOSE[5956] logger.c: Reliably Transmitting (NAT) to 66.114.80.25:1417: INVITE sip:000011206003@192.168.0.208:5060 SIP/2.0 Via: SIP/2.0/UDP 66.114.80.26:5060;branch=z9hG4bK5b691e06;rport From: ;tag=as2e186f49 To: ;tag=5B4AEC8-118A Contact: Call-ID: 537D0811-27FD11E0-8816CD5D-9D08DBFA@192.168.0.208 CSeq: 103 INVITE User-Agent: AcePBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 211 v=0 o=root 5946 5949 IN IP4 66.114.80.26 s=session c=IN IP4 66.114.80.26 t=0 0 m=audio 45294 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Jan 25 16:03:23] VERBOSE[5956] logger.c: <--- SIP read from 66.114.80.25:1417 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 66.114.80.26:5060;branch=z9hG4bK5b691e06;rport From: ;tag=as2e186f49 To: ;tag=5B4AEC8-118A Date: Tue, 25 Jan 2011 21:03:23 GMT Call-ID: 537D0811-27FD11E0-8816CD5D-9D08DBFA@192.168.0.208 Server: Cisco-SIPGateway/IOS-12.x CSeq: 103 INVITE Allow-Events: telephone-event Remote-Party-ID: ;party=called;screen=no;privacy=off Content-Length: 0 <-------------> [Jan 25 16:03:23] VERBOSE[5956] logger.c: --- (11 headers 0 lines) --- [Jan 25 16:03:23] DEBUG[5956] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '537D0811-27FD11E0-8816CD5D-9D08DBFA@192.168.0.208' Request 103: Found [Jan 25 16:03:23] VERBOSE[5956] logger.c: <--- SIP read from 66.114.80.25:1417 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 66.114.80.26:5060;branch=z9hG4bK5b691e06;rport From: ;tag=as2e186f49 To: ;tag=5B4AEC8-118A Date: Tue, 25 Jan 2011 21:03:23 GMT Call-ID: 537D0811-27FD11E0-8816CD5D-9D08DBFA@192.168.0.208 Server: Cisco-SIPGateway/IOS-12.x CSeq: 103 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER Supported: replaces Allow-Events: telephone-event Remote-Party-ID: ;party=called;screen=no;privacy=off Contact: Content-Type: application/sdp Content-Length: 249 v=0 o=CiscoSystemsSIP-GW-UserAgent 178 2856 IN IP4 192.168.0.208 s=SIP Call c=IN IP4 192.168.0.208 t=0 0 m=audio 16488 RTP/AVP 0 101 c=IN IP4 192.168.0.208 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 <-------------> [Jan 25 16:03:23] VERBOSE[5956] logger.c: --- (15 headers 11 lines) --- [Jan 25 16:03:23] DEBUG[5956] chan_sip.c: Acked pending invite 103 [Jan 25 16:03:23] DEBUG[5956] chan_sip.c: Stopping retransmission on '537D0811-27FD11E0-8816CD5D-9D08DBFA@192.168.0.208' of Request 103: Match Found [Jan 25 16:03:23] VERBOSE[5956] logger.c: Found RTP audio format 0 [Jan 25 16:03:23] VERBOSE[5956] logger.c: Found RTP audio format 101 [Jan 25 16:03:23] VERBOSE[5956] logger.c: Found audio description format PCMU for ID 0 [Jan 25 16:03:23] VERBOSE[5956] logger.c: Found audio description format telephone-event for ID 101 [Jan 25 16:03:23] VERBOSE[5956] logger.c: Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) [Jan 25 16:03:23] VERBOSE[5956] logger.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Jan 25 16:03:23] DEBUG[5956] chan_sip.c: Our T38 capability = (16160), peer T38 capability (16160), joint T38 capability (16160) [Jan 25 16:03:23] VERBOSE[5956] logger.c: Peer audio RTP is at port 192.168.0.208:16488 [Jan 25 16:03:23] DEBUG[5956] chan_sip.c: Peer doesn't provide T.38 UDPTL [Jan 25 16:03:23] DEBUG[5956] chan_sip.c: Strict routing enforced for session 537D0811-27FD11E0-8816CD5D-9D08DBFA@192.168.0.208 [Jan 25 16:03:23] VERBOSE[5956] logger.c: set_destination: Parsing for address/port to send to [Jan 25 16:03:23] VERBOSE[5956] logger.c: set_destination: set destination to 192.168.0.208, port 5060 [Jan 25 16:03:23] VERBOSE[5956] logger.c: Transmitting (NAT) to 66.114.80.25:1417: ACK sip:000011206003@192.168.0.208:5060 SIP/2.0 Via: SIP/2.0/UDP 66.114.80.26:5060;branch=z9hG4bK27d399dd;rport From: ;tag=as2e186f49 To: ;tag=5B4AEC8-118A Contact: Call-ID: 537D0811-27FD11E0-8816CD5D-9D08DBFA@192.168.0.208 CSeq: 103 ACK User-Agent: AcePBX Max-Forwards: 70 Content-Length: 0 --- [Jan 25 16:03:23] DEBUG[8237] rtp.c: RTP NAT: Got audio from other end. Now sending to address 66.114.80.25:16488 [Jan 25 16:03:25] VERBOSE[5956] logger.c: <--- SIP read from 66.114.80.25:52403 ---> INVITE sip:6469646701@66.114.80.26:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.208:5060;branch=z9hG4bK1017C5D From: ;tag=5B4AEC8-118A To: ;tag=as2e186f49 Date: Tue, 25 Jan 2011 21:03:25 GMT Call-ID: 537D0811-27FD11E0-8816CD5D-9D08DBFA@192.168.0.208 Supported: 100rel,timer,replaces Min-SE: 1800 Cisco-Guid: 1342662883-670896608-2282999133-2634603514 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER CSeq: 104 INVITE Max-Forwards: 10 Remote-Party-ID: ;party=calling;screen=no;privacy=off Timestamp: 1295989405 Contact: Expires: 180 Allow-Events: telephone-event Proxy-Authorization: Digest username="000011206003",realm="acepbx.com",uri="sip:6469646701@66.114.80.26:5060",response="28dd26fefb91468d95b8896ccfe54ca2",nonce="19c76cad",algorithm=MD5 Content-Type: application/sdp Content-Length: 398 v=0 o=CiscoSystemsSIP-GW-UserAgent 178 2857 IN IP4 192.168.0.208 s=SIP Call c=IN IP4 192.168.0.208 t=0 0 m=image 16488 udptl t38 c=IN IP4 192.168.0.208 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxFillBitRemoval:0 a=T38FaxTranscodingMMR:0 a=T38FaxTranscodingJBIG:0 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:200 a=T38FaxMaxDatagram:72 a=T38FaxUdpEC:t38UDPRedundancy <-------------> [Jan 25 16:03:25] VERBOSE[5956] logger.c: --- (21 headers 16 lines) --- [Jan 25 16:03:25] VERBOSE[5956] logger.c: Sending to 66.114.80.25 : 52403 (NAT) [Jan 25 16:03:25] VERBOSE[5956] logger.c: Got T.38 offer in SDP in dialog 537D0811-27FD11E0-8816CD5D-9D08DBFA@192.168.0.208 [Jan 25 16:03:25] VERBOSE[5956] logger.c: Capabilities: us - 0x4 (ulaw), peer - audio=0x0 (nothing)/video=0x0 (nothing), combined - 0x0 (nothing) [Jan 25 16:03:25] VERBOSE[5956] logger.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) [Jan 25 16:03:25] DEBUG[5956] chan_sip.c: Our T38 capability = (16160), peer T38 capability (16160), joint T38 capability (16160) [Jan 25 16:03:25] VERBOSE[5956] logger.c: Got T.38 Re-invite without audio. Keeping RTP active during T.38 session. [Jan 25 16:03:25] DEBUG[5956] chan_sip.c: Peer T.38 UDPTL is at port 192.168.0.208:16488 [Jan 25 16:03:25] VERBOSE[5956] logger.c: <--- Transmitting (NAT) to 66.114.80.25:52403 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.208:5060;branch=z9hG4bK1017C5D;received=66.114.80.25 From: ;tag=5B4AEC8-118A To: ;tag=as2e186f49 Call-ID: 537D0811-27FD11E0-8816CD5D-9D08DBFA@192.168.0.208 CSeq: 104 INVITE User-Agent: AcePBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: Content-Length: 0 <------------> [Jan 25 16:03:25] DEBUG[5956] chan_sip.c: T.38 UDPTL is at 66.114.80.26 port 40579 [Jan 25 16:03:25] DEBUG[5956] chan_sip.c: Our T38 capability (16160), peer T38 capability (16160), joint capability (16160) [Jan 25 16:03:25] VERBOSE[5956] logger.c: <--- Reliably Transmitting (NAT) to 66.114.80.25:52403 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.208:5060;branch=z9hG4bK1017C5D;received=66.114.80.25 From: ;tag=5B4AEC8-118A To: ;tag=as2e186f49 Call-ID: 537D0811-27FD11E0-8816CD5D-9D08DBFA@192.168.0.208 CSeq: 104 INVITE User-Agent: AcePBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: Content-Type: application/sdp Content-Length: 268 v=0 o=root 5946 5950 IN IP4 66.114.80.26 s=session c=IN IP4 66.114.80.26 t=0 0 m=image 40579 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:72 a=T38FaxMaxDatagram:72 a=T38FaxUdpEC:t38UDPRedundancy <------------> [Jan 25 16:03:25] DEBUG[5956] chan_sip.c: T38 state changed to 4 on channel SIP/000011206003-0000002a [Jan 25 16:03:25] DEBUG[8237] rtp.c: Got RTCP report of 72 bytes [Jan 25 16:03:25] VERBOSE[5956] logger.c: <--- SIP read from 66.114.80.25:52403 ---> ACK sip:6469646701@66.114.80.26:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.208:5060;branch=z9hG4bK10182107 From: ;tag=5B4AEC8-118A To: ;tag=as2e186f49 Date: Tue, 25 Jan 2011 21:03:25 GMT Call-ID: 537D0811-27FD11E0-8816CD5D-9D08DBFA@192.168.0.208 Max-Forwards: 10 CSeq: 104 ACK Proxy-Authorization: Digest username="000011206003",realm="acepbx.com",uri="sip:6469646701@66.114.80.26:5060",response="28dd26fefb91468d95b8896ccfe54ca2",nonce="19c76cad",algorithm=MD5 Content-Length: 0 <-------------> [Jan 25 16:03:25] VERBOSE[5956] logger.c: --- (10 headers 0 lines) --- [Jan 25 16:03:25] DEBUG[5956] chan_sip.c: Stopping retransmission on '537D0811-27FD11E0-8816CD5D-9D08DBFA@192.168.0.208' of Response 104: Match Found [Jan 25 16:03:25] DEBUG[8237] udptl.c: UDPTL NAT: Using address 66.114.80.25:16488 [Jan 25 16:03:26] DEBUG[8237] rtp.c: Got RTCP report of 132 bytes [Jan 25 16:03:28] VERBOSE[5956] logger.c: <--- SIP read from 66.114.80.25:52403 ---> REGISTER sip:pbxtest.acepbx.com:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.208:5060;branch=z9hG4bK101911FA From: "000011206003" ;tag=5B51E68-44C To: "000011206003" Date: Tue, 25 Jan 2011 21:03:28 GMT Call-ID: 42499FB3-27FA11E0-87E6CD5D-9D08DBFA User-Agent: Cisco-SIPGateway/IOS-12.x Max-Forwards: 10 Timestamp: 1295989408 CSeq: 29 REGISTER Contact: Expires: 120 Content-Length: 0 <-------------> [Jan 25 16:03:28] VERBOSE[5956] logger.c: --- (13 headers 0 lines) --- [Jan 25 16:03:28] DEBUG[5956] chan_sip.c: Allocating new SIP dialog for 42499FB3-27FA11E0-87E6CD5D-9D08DBFA - REGISTER (No RTP) [Jan 25 16:03:28] VERBOSE[5956] logger.c: Using latest REGISTER request as basis request [Jan 25 16:03:28] VERBOSE[5956] logger.c: Sending to 66.114.80.25 : 52403 (NAT) [Jan 25 16:03:28] VERBOSE[5956] logger.c: <--- Transmitting (NAT) to 66.114.80.25:52403 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.208:5060;branch=z9hG4bK101911FA;received=66.114.80.25 From: "000011206003" ;tag=5B51E68-44C To: "000011206003" Call-ID: 42499FB3-27FA11E0-87E6CD5D-9D08DBFA CSeq: 29 REGISTER User-Agent: AcePBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 <------------> [Jan 25 16:03:28] VERBOSE[5956] logger.c: <--- Transmitting (NAT) to 66.114.80.25:52403 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.0.208:5060;branch=z9hG4bK101911FA;received=66.114.80.25 From: "000011206003" ;tag=5B51E68-44C To: "000011206003" ;tag=as3a03482c Call-ID: 42499FB3-27FA11E0-87E6CD5D-9D08DBFA CSeq: 29 REGISTER User-Agent: AcePBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="acepbx.com", nonce="33d67f2f" Content-Length: 0 <------------> [Jan 25 16:03:28] VERBOSE[5956] logger.c: Scheduling destruction of SIP dialog '42499FB3-27FA11E0-87E6CD5D-9D08DBFA' in 32000 ms (Method: REGISTER) [Jan 25 16:03:28] VERBOSE[5956] logger.c: <--- SIP read from 66.114.80.25:52403 ---> REGISTER sip:pbxtest.acepbx.com:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.208:5060;branch=z9hG4bK101A14EB From: "000011206003" ;tag=5B51E68-44C To: "000011206003" Date: Tue, 25 Jan 2011 21:03:28 GMT Call-ID: 42499FB3-27FA11E0-87E6CD5D-9D08DBFA User-Agent: Cisco-SIPGateway/IOS-12.x Max-Forwards: 10 Timestamp: 1295989408 CSeq: 30 REGISTER Contact: Expires: 120 Authorization: Digest username="000011206003",realm="acepbx.com",uri="sip:pbxtest.acepbx.com:5060",response="ce619f5e65b9aa5470157aa627a9058c",nonce="33d67f2f",algorithm=MD5 Content-Length: 0 <-------------> [Jan 25 16:03:28] VERBOSE[5956] logger.c: --- (14 headers 0 lines) --- [Jan 25 16:03:28] VERBOSE[5956] logger.c: Using latest REGISTER request as basis request [Jan 25 16:03:28] VERBOSE[5956] logger.c: Sending to 66.114.80.25 : 52403 (NAT) [Jan 25 16:03:28] VERBOSE[5956] logger.c: <--- Transmitting (NAT) to 66.114.80.25:52403 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.208:5060;branch=z9hG4bK101A14EB;received=66.114.80.25 From: "000011206003" ;tag=5B51E68-44C To: "000011206003" Call-ID: 42499FB3-27FA11E0-87E6CD5D-9D08DBFA CSeq: 30 REGISTER User-Agent: AcePBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 <------------> [Jan 25 16:03:28] VERBOSE[5956] logger.c: <--- Transmitting (NAT) to 66.114.80.25:52403 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.208:5060;branch=z9hG4bK101A14EB;received=66.114.80.25 From: "000011206003" ;tag=5B51E68-44C To: "000011206003" ;tag=as3a03482c Call-ID: 42499FB3-27FA11E0-87E6CD5D-9D08DBFA CSeq: 30 REGISTER User-Agent: AcePBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Expires: 120 Contact: ;expires=120 Date: Tue, 25 Jan 2011 21:03:28 GMT Content-Length: 0 <------------> [Jan 25 16:03:28] VERBOSE[5956] logger.c: Scheduling destruction of SIP dialog '42499FB3-27FA11E0-87E6CD5D-9D08DBFA' in 32000 ms (Method: REGISTER) [Jan 25 16:03:31] DEBUG[8237] rtp.c: Got RTCP report of 88 bytes [Jan 25 16:03:35] DEBUG[5956] chan_sip.c: Auto destroying SIP dialog '89D0F9EC-27FA11E0-87E9CD5D-9D08DBFA' [Jan 25 16:03:35] VERBOSE[5956] logger.c: Really destroying SIP dialog '89D0F9EC-27FA11E0-87E9CD5D-9D08DBFA' Method: REGISTER [Jan 25 16:03:36] DEBUG[5956] chan_sip.c: Auto destroying SIP dialog '70d42e23493ff52b3b19df1921b16433@66.114.80.26' [Jan 25 16:03:36] VERBOSE[5956] logger.c: Really destroying SIP dialog '70d42e23493ff52b3b19df1921b16433@66.114.80.26' Method: REGISTER [Jan 25 16:03:38] DEBUG[8237] rtp.c: Got RTCP report of 88 bytes [Jan 25 16:03:40] DEBUG[5956] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - NOTIFY (No RTP) [Jan 25 16:03:40] VERBOSE[5956] logger.c: Scheduling destruction of SIP dialog '653f9d3c6a748d9e5690e225434fb221@66.114.80.26' in 32000 ms (Method: NOTIFY) [Jan 25 16:03:40] VERBOSE[5956] logger.c: Reliably Transmitting (NAT) to 66.114.80.25:52403: NOTIFY sip:000011206003@192.168.0.208:5060 SIP/2.0 Via: SIP/2.0/UDP 66.114.80.26:5060;branch=z9hG4bK219828c3;rport From: "Unknown" ;tag=as6b326220 To: Contact: Call-ID: 653f9d3c6a748d9e5690e225434fb221@66.114.80.26 CSeq: 102 NOTIFY User-Agent: AcePBX Max-Forwards: 70 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 95 Messages-Waiting: no Message-Account: sip:12128121210@66.114.80.26 Voice-Message: 0/0 (0/0) --- [Jan 25 16:03:40] VERBOSE[5956] logger.c: <--- SIP read from 66.114.80.25:1417 ---> SIP/2.0 481 Call Leg/Transaction Does Not Exist Via: SIP/2.0/UDP 66.114.80.26:5060;branch=z9hG4bK219828c3;rport From: "Unknown" ;tag=as6b326220 To: Date: Tue, 25 Jan 2011 21:03:40 GMT Call-ID: 653f9d3c6a748d9e5690e225434fb221@66.114.80.26 Server: Cisco-SIPGateway/IOS-12.x Content-Length: 0 CSeq: 102 NOTIFY <-------------> [Jan 25 16:03:40] VERBOSE[5956] logger.c: --- (9 headers 0 lines) --- [Jan 25 16:03:40] DEBUG[5956] chan_sip.c: Stopping retransmission on '653f9d3c6a748d9e5690e225434fb221@66.114.80.26' of Request 102: Match Found [Jan 25 16:03:40] VERBOSE[5956] logger.c: Really destroying SIP dialog '653f9d3c6a748d9e5690e225434fb221@66.114.80.26' Method: NOTIFY [Jan 25 16:03:41] DEBUG[8237] rtp.c: Got RTCP report of 88 bytes [Jan 25 16:03:44] DEBUG[8237] rtp.c: Got RTCP report of 88 bytes [Jan 25 16:03:49] DEBUG[8237] rtp.c: Got RTCP report of 88 bytes [Jan 25 16:03:53] VERBOSE[5956] logger.c: <--- SIP read from 66.114.83.7:5060 ---> BYE sip:7189282005@66.114.80.26:5060 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 66.114.83.7;branch=z9hG4bK0786.df242fd7.0 Via: SIP/2.0/UDP 66.114.76.201:5060;branch=z9hG4bK16396514D5 From: ;tag=7CD92148-202C To: "7189282005" ;tag=as57f27f73 Date: Tue, 25 Jan 2011 21:03:21 GMT Call-ID: 7d7cd6e9419a8b4217b6142c1631edeb@66.114.80.26 User-Agent: Cisco-SIPGateway/IOS-12.x Max-Forwards: 9 Timestamp: 1295989433 CSeq: 102 BYE Content-Length: 0 <-------------> [Jan 25 16:03:53] VERBOSE[5956] logger.c: --- (13 headers 0 lines) --- [Jan 25 16:03:53] VERBOSE[5956] logger.c: <--- Reliably Transmitting (NAT) to 66.114.83.7:5060 ---> SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 66.114.83.7;branch=z9hG4bK3786.05b76082.0;received=66.114.83.7 Via: SIP/2.0/UDP 66.114.76.201:5060;branch=z9hG4bK1639511725 From: ;tag=7CD92148-202C To: "7189282005" ;tag=as57f27f73 Call-ID: 7d7cd6e9419a8b4217b6142c1631edeb@66.114.80.26 CSeq: 101 INVITE User-Agent: AcePBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 <------------> [Jan 25 16:03:53] DEBUG[5956] chan_sip.c: Stopping retransmission on '7d7cd6e9419a8b4217b6142c1631edeb@66.114.80.26' of Response 101: Match Found [Jan 25 16:03:53] VERBOSE[5956] logger.c: Sending to 66.114.83.7 : 5060 (NAT) [Jan 25 16:03:53] VERBOSE[5956] logger.c: <--- Transmitting (NAT) to 66.114.83.7:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 66.114.83.7;branch=z9hG4bK0786.df242fd7.0;received=66.114.83.7 Via: SIP/2.0/UDP 66.114.76.201:5060;branch=z9hG4bK16396514D5 Record-Route: From: ;tag=7CD92148-202C To: "7189282005" ;tag=as57f27f73 Call-ID: 7d7cd6e9419a8b4217b6142c1631edeb@66.114.80.26 CSeq: 102 BYE User-Agent: AcePBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 <------------> [Jan 25 16:03:53] DEBUG[8237] channel.c: Didn't get a frame from channel: SIP/sipp1-0000002b [Jan 25 16:03:53] DEBUG[8237] channel.c: Bridge stops bridging channels SIP/000011206003-0000002a and SIP/sipp1-0000002b [Jan 25 16:03:53] DEBUG[8237] pbx.c: Function result is 'SIP' [Jan 25 16:03:53] DEBUG[8237] pbx.c: Launching 'NoOp' [Jan 25 16:03:53] VERBOSE[8237] logger.c: -- Executing [h@xyz:1] NoOp("SIP/000011206003-0000002a", "v2 hangup extension. CHANNELTYPE=SIP| CHANNEL=SIP/000011206003-0000002a") in new stack [Jan 25 16:03:53] DEBUG[8237] pbx.c: Function result is '1' [Jan 25 16:03:53] DEBUG[8237] pbx.c: Function result is '0' [Jan 25 16:03:53] DEBUG[8237] pbx.c: Expression result is '0' [Jan 25 16:03:53] DEBUG[8237] pbx.c: Launching 'GotoIf' [Jan 25 16:03:53] VERBOSE[8237] logger.c: -- Executing [h@xyz:2] GotoIf("SIP/000011206003-0000002a", "0?lines_done:+1") in new stack [Jan 25 16:03:53] VERBOSE[8237] logger.c: -- Goto (xyz,h,3) [Jan 25 16:03:53] DEBUG[8237] pbx.c: Function result is '1' [Jan 25 16:03:53] DEBUG[8237] pbx.c: Expression result is '1' [Jan 25 16:03:53] DEBUG[8237] pbx.c: Launching 'GotoIf' [Jan 25 16:03:53] VERBOSE[8237] logger.c: -- Executing [h@xyz:3] GotoIf("SIP/000011206003-0000002a", "1?lines_out:+1") in new stack [Jan 25 16:03:53] VERBOSE[8237] logger.c: -- Goto (xyz,h,6) [Jan 25 16:03:53] DEBUG[8237] pbx.c: Function result is '0' [Jan 25 16:03:53] DEBUG[8237] pbx.c: Expression result is '0' [Jan 25 16:03:53] DEBUG[8237] pbx.c: Launching 'GotoIf' [Jan 25 16:03:53] VERBOSE[8237] logger.c: -- Executing [h@xyz:6] GotoIf("SIP/000011206003-0000002a", "0?lines_done:+1") in new stack [Jan 25 16:03:53] VERBOSE[8237] logger.c: -- Goto (xyz,h,7) [Jan 25 16:03:53] DEBUG[8237] pbx.c: Function result is 'NOT_INUSE' [Jan 25 16:03:53] DEBUG[8237] pbx.c: Expression result is '0' [Jan 25 16:03:53] DEBUG[8237] pbx.c: Launching 'GotoIf' [Jan 25 16:03:53] VERBOSE[8237] logger.c: -- Executing [h@xyz:7] GotoIf("SIP/000011206003-0000002a", "0?+1:lines_done") in new stack [Jan 25 16:03:53] VERBOSE[8237] logger.c: -- Goto (xyz,h,9) [Jan 25 16:03:53] DEBUG[8237] pbx.c: Launching 'NoOp' [Jan 25 16:03:53] VERBOSE[8237] logger.c: -- Executing [h@xyz:9] NoOp("SIP/000011206003-0000002a", "") in new stack [Jan 25 16:03:53] DEBUG[8237] pbx.c: Function result is 'SIP' [Jan 25 16:03:53] DEBUG[8237] pbx.c: Expression result is '1' [Jan 25 16:03:53] DEBUG[8237] pbx.c: Launching 'GotoIf' [Jan 25 16:03:53] VERBOSE[8237] logger.c: -- Executing [h@xyz:10] GotoIf("SIP/000011206003-0000002a", "1?+1:uf_done") in new stack [Jan 25 16:03:53] VERBOSE[8237] logger.c: -- Goto (xyz,h,11) [Jan 25 16:03:53] DEBUG[8237] func_strings.c: FUNCTION REGEX (ZOMBIE)(SIP/000011206003-0000002a) [Jan 25 16:03:53] DEBUG[8237] pbx.c: Function result is '0' [Jan 25 16:03:53] DEBUG[8237] pbx.c: Expression result is '0' [Jan 25 16:03:53] DEBUG[8237] pbx.c: Launching 'GotoIf' [Jan 25 16:03:53] VERBOSE[8237] logger.c: -- Executing [h@xyz:11] GotoIf("SIP/000011206003-0000002a", "0?uf_done:+1") in new stack [Jan 25 16:03:53] VERBOSE[8237] logger.c: -- Goto (xyz,h,12) [Jan 25 16:03:53] DEBUG[8237] pbx.c: Function result is '0' [Jan 25 16:03:53] DEBUG[8237] pbx.c: Expression result is '0' [Jan 25 16:03:53] DEBUG[8237] pbx.c: Launching 'GotoIf' [Jan 25 16:03:53] VERBOSE[8237] logger.c: -- Executing [h@xyz:12] GotoIf("SIP/000011206003-0000002a", "0?uf_done:+1") in new stack [Jan 25 16:03:53] VERBOSE[8237] logger.c: -- Goto (xyz,h,13) [Jan 25 16:03:53] DEBUG[8237] pbx.c: Function result is 'v2;id=1295989380.42;root=1295989380.42;xfer=0;clid=7189282005;call=1:2:__1_4644>1:0:16469646701,ct=1,c=0' [Jan 25 16:03:53] DEBUG[8237] pbx.c: Function result is '0' [Jan 25 16:03:53] DEBUG[8237] pbx.c: Expression result is '0' [Jan 25 16:03:53] DEBUG[8237] pbx.c: Launching 'GotoIf' [Jan 25 16:03:53] VERBOSE[8237] logger.c: -- Executing [h@xyz:13] GotoIf("SIP/000011206003-0000002a", "0?uf_set:+1") in new stack [Jan 25 16:03:53] VERBOSE[8237] logger.c: -- Goto (xyz,h,14) [Jan 25 16:03:53] DEBUG[8237] pbx.c: Function result is '"v2;id=1295989380.42;root=1295989380.42;xfer=0;clid=7189282005;call=1:2:__1_4644>1:0:16469646701,ct=1,c=0"' [Jan 25 16:03:53] DEBUG[8237] pbx.c: Function result is 'v2;id=1295989380.42;root=1295989380.42;xfer=0;clid=7189282005;call=1:2:__1_4644>1:0:16469646701,ct=1,c=0' [Jan 25 16:03:53] DEBUG[8237] pbx.c: Function result is '"v2;id=1295989380.42;root=1295989380.42;xfer=0;clid=7189282005;call=1:2:__1_4644>1:0:16469646701,ct=1,c=0"' [Jan 25 16:03:53] DEBUG[8237] pbx.c: Expression result is '1' [Jan 25 16:03:53] DEBUG[8237] pbx.c: Launching 'GotoIf' [Jan 25 16:03:53] VERBOSE[8237] logger.c: -- Executing [h@xyz:14] GotoIf("SIP/000011206003-0000002a", "1?uf_done:+1") in new stack [Jan 25 16:03:53] VERBOSE[8237] logger.c: -- Goto (xyz,h,16) [Jan 25 16:03:53] DEBUG[8237] pbx.c: Launching 'NoOp' [Jan 25 16:03:53] VERBOSE[8237] logger.c: -- Executing [h@xyz:16] NoOp("SIP/000011206003-0000002a", "") in new stack [Jan 25 16:03:53] DEBUG[8237] pbx.c: Function result is 'Up' [Jan 25 16:03:53] DEBUG[8237] pbx.c: Launching 'NoOp' [Jan 25 16:03:53] VERBOSE[8237] logger.c: -- Executing [h@xyz:17] NoOp("SIP/000011206003-0000002a", "STATE=Up") in new stack [Jan 25 16:03:53] DEBUG[8237] pbx.c: Function result is 'Up' [Jan 25 16:03:53] DEBUG[8237] pbx.c: Expression result is '0' [Jan 25 16:03:53] DEBUG[8237] pbx.c: Launching 'GotoIf' [Jan 25 16:03:53] VERBOSE[8237] logger.c: -- Executing [h@xyz:18] GotoIf("SIP/000011206003-0000002a", "0?rqhangup_done:+1") in new stack [Jan 25 16:03:53] VERBOSE[8237] logger.c: -- Goto (xyz,h,19) [Jan 25 16:03:53] DEBUG[8237] pbx.c: Function result is 'Up' [Jan 25 16:03:53] DEBUG[8237] pbx.c: Expression result is '0' [Jan 25 16:03:53] DEBUG[8237] pbx.c: Launching 'GotoIf' [Jan 25 16:03:53] VERBOSE[8237] logger.c: -- Executing [h@xyz:19] GotoIf("SIP/000011206003-0000002a", "0?rqhangup_done:+1") in new stack [Jan 25 16:03:53] VERBOSE[8237] logger.c: -- Goto (xyz,h,20) [Jan 25 16:03:53] DEBUG[8237] pbx.c: Launching 'ImportVar' [Jan 25 16:03:53] VERBOSE[8237] logger.c: -- Executing [h@xyz:20] ImportVar("SIP/000011206003-0000002a", "PEER_C_URI=SIP/sipp1-0000002b|EXT_CONTACT_URI") in new stack [Jan 25 16:03:53] DEBUG[8237] pbx.c: Function result is 'Up' [Jan 25 16:03:53] DEBUG[8237] pbx.c: Function result is '"root_id=1295989380.42&bridge_peer=SIP/sipp1-0000002b&state=Up&peer_c_uri=sip:000016469646701@66.114.76.201:5060"' [Jan 25 16:03:53] DEBUG[8237] pbx.c: Launching 'Set' [Jan 25 16:03:53] VERBOSE[8237] logger.c: -- Executing [h@xyz:21] Set("SIP/000011206003-0000002a", "PARAMS="root_id=1295989380.42&bridge_peer=SIP/sipp1-0000002b&state=Up&peer_c_uri=sip:000016469646701@66.114.76.201:5060"") in new stack [Jan 25 16:03:53] DEBUG[8237] pbx.c: Launching 'AceCheckHangup' [Jan 25 16:03:53] VERBOSE[8237] logger.c: -- Executing [h@xyz:22] AceCheckHangup("SIP/000011206003-0000002a", "") in new stack [Jan 25 16:03:53] VERBOSE[8237] logger.c: -- AceCheckHangup: SIP/000011206003-0000002a is alive [Jan 25 16:03:53] DEBUG[8237] pbx.c: Expression result is '1' [Jan 25 16:03:53] DEBUG[8237] pbx.c: Launching 'GotoIf' [Jan 25 16:03:53] VERBOSE[8237] logger.c: -- Executing [h@xyz:23] GotoIf("SIP/000011206003-0000002a", "1?+1:deadagi") in new stack [Jan 25 16:03:53] VERBOSE[8237] logger.c: -- Goto (xyz,h,24) [Jan 25 16:03:53] DEBUG[8237] pbx.c: Launching 'AGI' [Jan 25 16:03:53] VERBOSE[8237] logger.c: -- Executing [h@xyz:24] AGI("SIP/000011206003-0000002a", "agi://pbxtest/HANGUP?root_id=1295989380.42&bridge_peer=SIP/sipp1-0000002b&state=Up&peer_c_uri=sip:000016469646701@66.114.76.201:5060") in new stack [Jan 25 16:03:53] VERBOSE[5956] logger.c: <--- SIP read from 66.114.83.7:5060 ---> ACK sip:7189282005@66.114.80.26:5060 SIP/2.0 Via: SIP/2.0/UDP 66.114.83.7;branch=z9hG4bK3786.05b76082.0 From: ;tag=7CD92148-202C Call-ID: 7d7cd6e9419a8b4217b6142c1631edeb@66.114.80.26 To: "7189282005" ;tag=as57f27f73 CSeq: 101 ACK Route: Content-Length: 0 <-------------> [Jan 25 16:03:53] VERBOSE[5956] logger.c: --- (8 headers 0 lines) --- [Jan 25 16:03:53] DEBUG[5956] chan_sip.c: Stopping retransmission on '7d7cd6e9419a8b4217b6142c1631edeb@66.114.80.26' of Response 101: Match Not Found [Jan 25 16:03:53] DEBUG[8237] pbx.c: Function result is 'v2;id=1295989380.42;root=1295989380.42;xfer=0;clid=7189282005;call=1:2:__1_4644>1:0:16469646701,ct=1,c=0' [Jan 25 16:03:53] DEBUG[8237] pbx.c: Function result is '1' [Jan 25 16:03:53] DEBUG[8237] pbx.c: Function result is '527' [Jan 25 16:03:53] DEBUG[8237] pbx.c: Function result is '0' [Jan 25 16:03:53] DEBUG[8237] pbx.c: Function result is '0' [Jan 25 16:03:53] DEBUG[8237] pbx.c: Function result is '520' [Jan 25 16:03:53] DEBUG[8237] pbx.c: Function result is '0' [Jan 25 16:03:53] DEBUG[8237] pbx.c: Function result is '2' [Jan 25 16:03:53] DEBUG[8237] pbx.c: Function result is '1920' [Jan 25 16:03:53] DEBUG[8237] pbx.c: Function result is '5' [Jan 25 16:03:53] DEBUG[8237] pbx.c: Function result is '1' [Jan 25 16:03:53] DEBUG[8237] pbx.c: Function result is '527' [Jan 25 16:03:53] DEBUG[8237] pbx.c: Function result is '0' [Jan 25 16:03:53] DEBUG[8237] pbx.c: Function result is '1' [Jan 25 16:03:53] VERBOSE[8237] logger.c: -- AGI Script agi://pbxtest/HANGUP?root_id=1295989380.42&bridge_peer=SIP/sipp1-0000002b&state=Up&peer_c_uri=sip:000016469646701@66.114.76.201:5060 completed, returning 0 [Jan 25 16:03:53] DEBUG[8237] pbx.c: Launching 'Goto' [Jan 25 16:03:53] VERBOSE[8237] logger.c: -- Executing [h@xyz:25] Goto("SIP/000011206003-0000002a", "rqhangup_done") in new stack [Jan 25 16:03:53] VERBOSE[8237] logger.c: -- Goto (xyz,h,27) [Jan 25 16:03:53] DEBUG[8237] pbx.c: Launching 'NoOp' [Jan 25 16:03:53] VERBOSE[8237] logger.c: -- Executing [h@xyz:27] NoOp("SIP/000011206003-0000002a", "") in new stack [Jan 25 16:03:53] DEBUG[8237] channel.c: Hanging up channel 'SIP/sipp1-0000002b' [Jan 25 16:03:53] DEBUG[8237] chan_sip.c: Hangup call SIP/sipp1-0000002b, SIP callid 7d7cd6e9419a8b4217b6142c1631edeb@66.114.80.26) [Jan 25 16:03:53] DEBUG[8237] rtp.c: Channel '' has no RTP, not doing anything [Jan 25 16:03:53] DEBUG[8237] app_dial.c: Exiting with DIALSTATUS=ANSWER. [Jan 25 16:03:53] VERBOSE[8237] logger.c: -- AGI Script Executing Application: (Set) Options: (DEVSTATE(Custom:LINES_COMPANY_2_STATE)=NOT_INUSE) [Jan 25 16:03:53] VERBOSE[5956] logger.c: Really destroying SIP dialog '7d7cd6e9419a8b4217b6142c1631edeb@66.114.80.26' Method: BYE [Jan 25 16:03:53] VERBOSE[8237] logger.c: -- AGI Script Executing Application: (Set) Options: (GROUP()=none@LINES_COMPANY_2_OUT) [Jan 25 16:03:53] VERBOSE[8237] logger.c: -- AGI Script agi://pbxtest/CP?blnd_xfer= completed, returning 0 [Jan 25 16:03:53] DEBUG[8237] pbx.c: Launching 'Hangup' [Jan 25 16:03:53] VERBOSE[8237] logger.c: -- Executing [6469646701@xyz:4] Hangup("SIP/000011206003-0000002a", "") in new stack [Jan 25 16:03:53] DEBUG[8237] pbx.c: Spawn extension (xyz,6469646701,4) exited non-zero on 'SIP/000011206003-0000002a' [Jan 25 16:03:53] VERBOSE[8237] logger.c: == Spawn extension (xyz, 6469646701, 4) exited non-zero on 'SIP/000011206003-0000002a' [Jan 25 16:03:53] DEBUG[8237] channel.c: Soft-Hanging up channel 'SIP/000011206003-0000002a' [Jan 25 16:03:53] DEBUG[8237] channel.c: Hanging up channel 'SIP/000011206003-0000002a' [Jan 25 16:03:53] DEBUG[8237] chan_sip.c: Hangup call SIP/000011206003-0000002a, SIP callid 537D0811-27FD11E0-8816CD5D-9D08DBFA@192.168.0.208) [Jan 25 16:03:53] DEBUG[8237] chan_sip.c: update_call_counter(000011206003) - decrement call limit counter on hangup [Jan 25 16:03:53] DEBUG[8237] chan_sip.c: Call from peer '000011206003' removed from call limit 3 [Jan 25 16:03:53] VERBOSE[8237] logger.c: Scheduling destruction of SIP dialog '537D0811-27FD11E0-8816CD5D-9D08DBFA@192.168.0.208' in 32000 ms (Method: ACK) [Jan 25 16:03:53] DEBUG[8237] chan_sip.c: Strict routing enforced for session 537D0811-27FD11E0-8816CD5D-9D08DBFA@192.168.0.208 [Jan 25 16:03:53] VERBOSE[8237] logger.c: set_destination: Parsing for address/port to send to [Jan 25 16:03:53] VERBOSE[8237] logger.c: set_destination: set destination to 192.168.0.208, port 5060 [Jan 25 16:03:53] VERBOSE[8237] logger.c: Reliably Transmitting (NAT) to 66.114.80.25:52403: BYE sip:000011206003@192.168.0.208:5060 SIP/2.0 Via: SIP/2.0/UDP 66.114.80.26:5060;branch=z9hG4bK56bbb8b7;rport From: ;tag=as2e186f49 To: ;tag=5B4AEC8-118A Call-ID: 537D0811-27FD11E0-8816CD5D-9D08DBFA@192.168.0.208 CSeq: 104 BYE User-Agent: AcePBX Max-Forwards: 70 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [Jan 25 16:03:53] VERBOSE[5956] logger.c: <--- SIP read from 66.114.80.25:1417 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 66.114.80.26:5060;branch=z9hG4bK56bbb8b7;rport From: ;tag=as2e186f49 To: ;tag=5B4AEC8-118A Date: Tue, 25 Jan 2011 21:03:53 GMT Call-ID: 537D0811-27FD11E0-8816CD5D-9D08DBFA@192.168.0.208 Server: Cisco-SIPGateway/IOS-12.x Content-Length: 0 CSeq: 104 BYE <-------------> [Jan 25 16:03:53] VERBOSE[5956] logger.c: --- (9 headers 0 lines) --- [Jan 25 16:03:53] DEBUG[5956] chan_sip.c: Stopping retransmission on '537D0811-27FD11E0-8816CD5D-9D08DBFA@192.168.0.208' of Request 104: Match Found [Jan 25 16:03:53] VERBOSE[5956] logger.c: SIP Response message for INCOMING dialog BYE arrived [Jan 25 16:03:53] VERBOSE[5956] logger.c: Really destroying SIP dialog '537D0811-27FD11E0-8816CD5D-9D08DBFA@192.168.0.208' Method: ACK