SIP Debugging Enabled for IP: 216.229.127.55 Oct 4 14:36:15 DEBUG[1671]: chan_sip.c:7354 handle_request: Check for res for 5305715501.pw.digitalpath.net Oct 4 14:36:15 DEBUG[1671]: chan_sip.c:1623 update_user_counter: Call from user '5305715501.pw.digitalpath.net' is 1 out of 0 Oct 4 14:36:15 DEBUG[1671]: chan_sip.c:4643 build_route: build_route: Contact hop: Ray Van Dolson Oct 4 14:36:15 DEBUG[1671]: pbx.c:1194 pbx_substitute_variables_helper: Expression is '0' Oct 4 14:36:15 DEBUG[1671]: pbx.c:1274 pbx_extension_helper: Launching 'GotoIf' -- Executing GotoIf("SIP/5305715501.pw.digitalpath.net-48fb", "0?2:4") in new stack -- Goto (from-sip,8997787,4) Oct 4 14:36:15 DEBUG[1671]: pbx.c:1274 pbx_extension_helper: Launching 'Macro' -- Executing Macro("SIP/5305715501.pw.digitalpath.net-48fb", "dial-local2|8997787|530") in new stack Oct 4 14:36:15 DEBUG[1671]: pbx.c:1274 pbx_extension_helper: Launching 'ChanIsAvail' -- Executing ChanIsAvail("SIP/5305715501.pw.digitalpath.net-48fb", "SIP/5308997787.pw.digitalpath.net") in new stack Urgent handler Oct 4 14:36:15 WARNING[1671]: chan_sip.c:1401 create_addr: No such host: 5308997787.pw.digitalpath.net Oct 4 14:36:15 DEBUG[1671]: pbx.c:1274 pbx_extension_helper: Launching 'MailboxExists' -- Executing MailboxExists("SIP/5305715501.pw.digitalpath.net-48fb", "5308997787") in new stack Oct 4 14:36:15 DEBUG[1671]: pbx.c:1274 pbx_extension_helper: Launching 'NoOp' -- Executing NoOp("SIP/5305715501.pw.digitalpath.net-48fb", "Hello") in new stack Oct 4 14:36:15 DEBUG[1671]: pbx.c:1274 pbx_extension_helper: Launching 'MYSQL' -- Executing MYSQL("SIP/5305715501.pw.digitalpath.net-48fb", "Connect connid localhost asterisk astpw123 asterisk") in new stack Urgent handler Oct 4 14:36:15 DEBUG[1671]: pbx.c:1274 pbx_extension_helper: Launching 'MYSQL' -- Executing MYSQL("SIP/5305715501.pw.digitalpath.net-48fb", "Query resultid 1 SELECT SUBSTRING(USERS_USERNAME, 1, 10) AS SIP_NUMBER, SUBSTRING(USERS_USERNAME, 12) AS SIP_DOMAIN FROM AST_LNP LEFT JOIN AST_USERS ON LNP_USERS_ID = USERS_ID WHERE LNP_NUMBER = '5308997787' LIMIT 1") in new stack Urgent handler Oct 4 14:36:15 DEBUG[1671]: pbx.c:1274 pbx_extension_helper: Launching 'MYSQL' -- Executing MYSQL("SIP/5305715501.pw.digitalpath.net-48fb", "Fetch foundRow 2 SIP_Num SIP_Domain") in new stack Oct 4 14:36:15 DEBUG[1671]: pbx.c:1274 pbx_extension_helper: Launching 'MYSQL' -- Executing MYSQL("SIP/5305715501.pw.digitalpath.net-48fb", "Clear 2") in new stack Oct 4 14:36:15 DEBUG[1671]: pbx.c:1274 pbx_extension_helper: Launching 'MYSQL' -- Executing MYSQL("SIP/5305715501.pw.digitalpath.net-48fb", "Disconnect 1") in new stack Oct 4 14:36:15 DEBUG[1671]: pbx.c:1194 pbx_substitute_variables_helper: Expression is '0' Oct 4 14:36:15 DEBUG[1671]: pbx.c:1274 pbx_extension_helper: Launching 'GotoIf' -- Executing GotoIf("SIP/5305715501.pw.digitalpath.net-48fb", "0?200:120") in new stack -- Goto (macro-dial-local2,s,120) Oct 4 14:36:15 DEBUG[1671]: pbx.c:1274 pbx_extension_helper: Launching 'Macro' -- Executing Macro("SIP/5305715501.pw.digitalpath.net-48fb", "dial-provider|5308997787") in new stack Oct 4 14:36:15 DEBUG[1671]: pbx.c:1274 pbx_extension_helper: Launching 'Dial' -- Executing Dial("SIP/5305715501.pw.digitalpath.net-48fb", "SIP/5308997787@pacwest-outbound||tg") in new stack Oct 4 14:36:15 DEBUG[1671]: app_dial.c:509 dial_exec: SIMPLE DIAL (NO URL) Oct 4 14:36:15 DEBUG[1671]: chan_sip.c:2363 sip_alloc: Allocating new SIP call for (null) Oct 4 14:36:15 DEBUG[1671]: chan_sip.c:1310 create_addr: Setting NAT on RTP to 0 Oct 4 14:36:15 DEBUG[1671]: chan_sip.c:1490 sip_call: Outgoing Call for 5308997787 Oct 4 14:36:15 DEBUG[1671]: chan_sip.c:1595 update_user_counter: 5308997787 is not a local user We're at 208.53.92.4 port 22880 Answering/Requesting with root capability 0x4 (ulaw) Answering with preferred capability 0x8 (alaw) Answering with non-codec capability 0x1 (telephone-event) 12 headers, 11 lines Reliably Transmitting: INVITE sip:5308997787@216.229.127.55 SIP/2.0 Via: SIP/2.0/UDP 208.53.92.4:5060;branch=z9hG4bK0b9240bb From: "DIGITAL PATH" ;tag=as53a24f4b To: Contact: Call-ID: 314856d01c16022c0bfb02fd2817548e@208.53.92.4 CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Tue, 04 Oct 2005 21:36:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 236 v=0 o=root 1671 1671 IN IP4 208.53.92.4 s=session c=IN IP4 208.53.92.4 t=0 0 m=audio 22880 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=silenceSupp:off - - - - (no NAT) to 216.229.127.55:5060 -- Called 5308997787@pacwest-outbound Oct 4 14:36:15 DEBUG[1671]: channel.c:1752 ast_set_read_format: Set channel SIP/pacwest-outbound-1f74 to read format ulaw Oct 4 14:36:15 DEBUG[1671]: channel.c:1719 ast_set_write_format: Set channel SIP/5305715501.pw.digitalpath.net-48fb to write format ulaw Oct 4 14:36:15 DEBUG[1671]: channel.c:1719 ast_set_write_format: Set channel SIP/pacwest-outbound-1f74 to write format ulaw Oct 4 14:36:15 DEBUG[1671]: channel.c:1752 ast_set_read_format: Set channel SIP/5305715501.pw.digitalpath.net-48fb to read format ulaw Urgent handler Urgent handler Sip read: SIP/2.0 100 Try Via: SIP/2.0/UDP 208.53.92.4:5060;branch=z9hG4bK0b9240bb To: sip:5308997787@216.229.127.55 From: "DIGITAL PATH" ;tag=as53a24f4b Call-ID: 314856d01c16022c0bfb02fd2817548e@208.53.92.4 CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Tue, 04 Oct 2005 21:36:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 0 11 headers, 0 lines Oct 4 14:36:16 DEBUG[1671]: chan_sip.c:873 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '314856d01c16022c0bfb02fd2817548e@208.53.92.4' Request 102: Found Urgent handler Oct 4 14:36:17 DEBUG[1671]: rtp.c:375 ast_rtcp_read: Got RTCP report of 44 bytes Sip read: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 208.53.92.4:5060;branch=z9hG4bK0b9240bb To: sip:5308997787@216.229.127.55;tag=1c1635606641 From: "DIGITAL PATH" ;tag=as53a24f4b Call-ID: 314856d01c16022c0bfb02fd2817548e@208.53.92.4 CSeq: 102 INVITE Contact: sip:99925@66.53.126.36 Record-Route: Supported: em, timer, replaces, path Allow: REGISTER, OPTIONS, INVITE, ACK, CANCEL, BYE, NOTIFY, PRACK, REFER, INFO, SUBSCRIBE, UPDATE Content-Type: application/sdp Content-Length: 261 v=0 o=CNI_Gateway 1635692441 1635692136 IN IP4 66.53.126.36 s=Phone-Call c=IN IP4 66.53.126.36 t=0 0 m=audio 9440 RTP/AVP 0 106 101 a=rtpmap:0 pcmu/8000 a=rtpmap:106 X-NSE/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv 12 headers, 12 lines Oct 4 14:36:17 DEBUG[1671]: chan_sip.c:873 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '314856d01c16022c0bfb02fd2817548e@208.53.92.4' Request 102: Found Found RTP audio format 0 Found RTP audio format 106 Found RTP audio format 101 Peer audio RTP is at port 66.53.126.36:9440 Oct 4 14:36:17 DEBUG[1671]: chan_sip.c:2729 process_sdp: Peer audio RTP is at port 66.53.126.36:9440 Found description format pcmu Found description format X-NSE Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) Urgent handler -- SIP/pacwest-outbound-1f74 is making progress passing it to SIP/5305715501.pw.digitalpath.net-48fb Urgent handler Oct 4 14:36:17 DEBUG[1671]: rtp.c:1195 ast_rtp_write: Ooh, format changed from unknown to ulaw Oct 4 14:36:17 DEBUG[1671]: rtp.c:1195 ast_rtp_write: Ooh, format changed from unknown to ulaw Oct 4 14:36:20 NOTICE[1671]: rtp.c:509 ast_rtp_read: Unknown RTP codec 100 received Oct 4 14:36:28 DEBUG[1671]: chan_sip.c:2363 sip_alloc: Allocating new SIP call for 7632f070-3c817da0@65.164.104.108 Oct 4 14:36:28 DEBUG[1671]: chan_sip.c:2363 sip_alloc: Allocating new SIP call for 1e76d9f0-7c32a220@65.164.104.108 Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 208.53.92.4:5060;branch=z9hG4bK0b9240bb To: sip:5308997787@216.229.127.55;tag=1c1635606641 From: "DIGITAL PATH" ;tag=as53a24f4b Call-ID: 314856d01c16022c0bfb02fd2817548e@208.53.92.4 CSeq: 102 INVITE Contact: sip:99925@66.53.126.36 Record-Route: Supported: em, timer, replaces, path Allow: REGISTER, OPTIONS, INVITE, ACK, CANCEL, BYE, NOTIFY, PRACK, REFER, INFO, SUBSCRIBE, UPDATE Content-Type: application/sdp Content-Length: 261 v=0 o=CNI_Gateway 1635692441 1635692136 IN IP4 66.53.126.36 s=Phone-Call c=IN IP4 66.53.126.36 t=0 0 m=audio 9440 RTP/AVP 0 106 101 a=rtpmap:0 pcmu/8000 a=rtpmap:106 X-NSE/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv 12 headers, 12 lines Oct 4 14:36:28 DEBUG[1671]: chan_sip.c:823 __sip_ack: Acked pending invite 102 Oct 4 14:36:28 DEBUG[1671]: chan_sip.c:841 __sip_ack: Stopping retransmission on '314856d01c16022c0bfb02fd2817548e@208.53.92.4' of Request 102: Found Found RTP audio format 0 Found RTP audio format 106 Found RTP audio format 101 Peer audio RTP is at port 66.53.126.36:9440 Oct 4 14:36:28 DEBUG[1671]: chan_sip.c:2729 process_sdp: Peer audio RTP is at port 66.53.126.36:9440 Found description format pcmu Found description format X-NSE Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) Oct 4 14:36:28 DEBUG[1671]: chan_sip.c:4618 build_route: build_route: Record-Route hop: Oct 4 14:36:28 DEBUG[1671]: chan_sip.c:4643 build_route: build_route: Contact hop: sip:99925@66.53.126.36 list_route: hop: list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 216.229.127.55, port 5060 Transmitting: ACK sip:5308997787@216.229.127.55 SIP/2.0 Via: SIP/2.0/UDP 208.53.92.4:5060;branch=z9hG4bK566f16e4 Route: From: "DIGITAL PATH" ;tag=as53a24f4b To: ;tag=1c1635606641 Contact: Call-ID: 314856d01c16022c0bfb02fd2817548e@208.53.92.4 CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 216.229.127.55:5060 Urgent handler -- SIP/pacwest-outbound-1f74 answered SIP/5305715501.pw.digitalpath.net-48fb Urgent handler Oct 4 14:36:28 DEBUG[1671]: channel.c:1752 ast_set_read_format: Set channel SIP/5305715501.pw.digitalpath.net-48fb to read format ulaw Oct 4 14:36:28 DEBUG[1671]: channel.c:1719 ast_set_write_format: Set channel SIP/pacwest-outbound-1f74 to write format ulaw Oct 4 14:36:28 DEBUG[1671]: channel.c:1719 ast_set_write_format: Set channel SIP/5305715501.pw.digitalpath.net-48fb to write format ulaw Oct 4 14:36:28 DEBUG[1671]: channel.c:1752 ast_set_read_format: Set channel SIP/pacwest-outbound-1f74 to read format ulaw Oct 4 14:36:28 DEBUG[1671]: chan_sip.c:1822 sip_answer: sip_answer(SIP/5305715501.pw.digitalpath.net-48fb) -- Attempting native bridge of SIP/5305715501.pw.digitalpath.net-48fb and SIP/pacwest-outbound-1f74 Urgent handler Oct 4 14:36:28 DEBUG[1671]: chan_sip.c:841 __sip_ack: Stopping retransmission on 'fba96271-72dcf2f9@65.164.104.108' of Response 102: Found Sip read: INVITE sip:5305715501@208.53.92.4 SIP/2.0 Via: SIP/2.0/UDP 216.229.127.55:5060;branch=z9hG4bK2a54b999b7a-97e1ba8a Via: SIP/2.0/UDP 66.53.126.36;branch=z9hG4bKac1867396736 To: "DIGITAL PATH" ;tag=as53a24f4b From: sip:5308997787@216.229.127.55;tag=1c1635606641 Call-ID: 314856d01c16022c0bfb02fd2817548e@208.53.92.4 CSeq: 1 INVITE Max-Forwards: 69 Contact: sip:99925@66.53.126.36 Record-Route: Supported: em, timer, replaces, path Allow: REGISTER, OPTIONS, INVITE, ACK, CANCEL, BYE, NOTIFY, PRACK, REFER, INFO, SUBSCRIBE, UPDATE Content-Type: application/sdp Content-Length: 292 v=0 o=CNI_Gateway 1635692441 1635692137 IN IP4 66.53.126.36 s=Phone-Call c=IN IP4 66.53.126.36 t=0 0 m=image 9442 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxMaxBuffer:1024 a=T38FaxMaxDatagram:238 a=T38FaxRateManagement:transferredTCF a=T38FaxUdpEC:t38UDPRedundancy 14 headers, 12 lines Using latest request as basis request Sending to 216.229.127.55 : 5060 (non-NAT) Oct 4 14:36:30 WARNING[1671]: chan_sip.c:2712 process_sdp: Unknown SDP media type in offer image 9442 udptl t38 Transmitting (no NAT): SIP/2.0 488 Not acceptable here Via: SIP/2.0/UDP 216.229.127.55:5060;branch=z9hG4bK2a54b999b7a-97e1ba8a Via: SIP/2.0/UDP 66.53.126.36;branch=z9hG4bKac1867396736 From: sip:5308997787@216.229.127.55;tag=1c1635606641 To: "DIGITAL PATH" ;tag=as53a24f4b Call-ID: 314856d01c16022c0bfb02fd2817548e@208.53.92.4 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 216.229.127.55:5060 Urgent handler Sip read: ACK sip:5305715501@208.53.92.4 SIP/2.0 Via: SIP/2.0/UDP 216.229.127.55:5060;branch=z9hG4bK2a54b999b7a-97e1ba8a To: "DIGITAL PATH" ;tag=as53a24f4b From: sip:5308997787@216.229.127.55;tag=1c1635606641 Call-ID: 314856d01c16022c0bfb02fd2817548e@208.53.92.4 CSeq: 1 ACK User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 9 headers, 0 lines Oct 4 14:36:42 DEBUG[1671]: chan_sip.c:772 __sip_autodestruct: Auto destroying call 'a84d0e2f-ce6c6834@172.25.163.2' Oct 4 14:36:43 DEBUG[1671]: chan_sip.c:2363 sip_alloc: Allocating new SIP call for 7632f070-3c817da0@65.164.104.108 Oct 4 14:36:43 DEBUG[1671]: chan_sip.c:2363 sip_alloc: Allocating new SIP call for 1e76d9f0-7c32a220@65.164.104.108 Oct 4 14:36:50 DEBUG[1671]: chan_sip.c:2363 sip_alloc: Allocating new SIP call for (null) Oct 4 14:36:50 DEBUG[1671]: chan_sip.c:2363 sip_alloc: Allocating new SIP call for (null) Oct 4 14:36:58 DEBUG[1671]: chan_sip.c:2363 sip_alloc: Allocating new SIP call for 7632f070-3c817da0@65.164.104.108 Oct 4 14:36:58 DEBUG[1671]: chan_sip.c:2363 sip_alloc: Allocating new SIP call for 1e76d9f0-7c32a220@65.164.104.108 Oct 4 14:36:59 DEBUG[1671]: channel.c:2675 ast_channel_bridge: Didn't get a frame from channel: SIP/5305715501.pw.digitalpath.net-48fb Oct 4 14:36:59 DEBUG[1671]: channel.c:2746 ast_channel_bridge: Bridge stops bridging channels SIP/5305715501.pw.digitalpath.net-48fb and SIP/pacwest-outbound-1f74 Oct 4 14:36:59 DEBUG[1671]: channel.c:739 ast_hangup: Hanging up channel 'SIP/pacwest-outbound-1f74' Oct 4 14:36:59 DEBUG[1671]: chan_sip.c:1711 sip_hangup: sip_hangup(SIP/pacwest-outbound-1f74) Oct 4 14:36:59 DEBUG[1671]: chan_sip.c:1726 sip_hangup: update_user_counter(5308997787) - decrement inUse counter Oct 4 14:36:59 DEBUG[1671]: chan_sip.c:1595 update_user_counter: 5308997787 is not a local user set_destination: Parsing for address/port to send to set_destination: set destination to 216.229.127.55, port 5060 Reliably Transmitting: BYE sip:99925@66.53.126.36 SIP/2.0 Via: SIP/2.0/UDP 208.53.92.4:5060;branch=z9hG4bK47dfe7eb Route: From: "DIGITAL PATH" ;tag=as53a24f4b To: sip:5308997787@216.229.127.55;tag=1c1635606641 Contact: Call-ID: 314856d01c16022c0bfb02fd2817548e@208.53.92.4 CSeq: 103 BYE User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 216.229.127.55:5060 Oct 4 14:36:59 DEBUG[1671]: app_dial.c:1054 dial_exec: Exiting with DIALSTATUS=ANSWER. Oct 4 14:36:59 DEBUG[1671]: app_macro.c:159 macro_exec: Spawn extension (macro-dial-provider,s,1) exited non-zero on 'SIP/5305715501.pw.digitalpath.net-48fb' in macro 'dial-provider' Oct 4 14:36:59 DEBUG[1671]: app_macro.c:159 macro_exec: Spawn extension (macro-dial-local2,s,120) exited non-zero on 'SIP/5305715501.pw.digitalpath.net-48fb' in macro 'dial-local2' Oct 4 14:36:59 DEBUG[1671]: pbx.c:1851 ast_pbx_run: Spawn extension (from-sip,8997787,4) exited non-zero on 'SIP/5305715501.pw.digitalpath.net-48fb' Oct 4 14:36:59 DEBUG[1671]: pbx.c:1274 pbx_extension_helper: Launching 'Hangup' -- Executing Hangup("SIP/5305715501.pw.digitalpath.net-48fb", "") in new stack Oct 4 14:36:59 DEBUG[1671]: pbx.c:1971 ast_pbx_run: Spawn extension (from-sip,h,1) exited non-zero on 'SIP/5305715501.pw.digitalpath.net-48fb' Urgent handler Oct 4 14:36:59 DEBUG[1671]: cdr_addon_mysql.c:178 mysql_log: cdr_mysql: inserting a CDR record. Oct 4 14:36:59 DEBUG[1671]: cdr_addon_mysql.c:197 mysql_log: cdr_mysql: SQL command as follows: INSERT INTO cdr (calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode) VALUES ('2005-10-04 14:36:15','DIGITAL PATH <5305715501>','5305715501','8997787','from-sip', 'SIP/5305715501.pw.digitalpath.net-48fb','SIP/pacwest-outbound-1f74','Hangup','',44,31,'ANSWERED',3,'') Oct 4 14:36:59 DEBUG[1671]: channel.c:739 ast_hangup: Hanging up channel 'SIP/5305715501.pw.digitalpath.net-48fb' Oct 4 14:36:59 DEBUG[1671]: chan_sip.c:1711 sip_hangup: sip_hangup(SIP/5305715501.pw.digitalpath.net-48fb) Oct 4 14:36:59 DEBUG[1671]: chan_sip.c:1726 sip_hangup: update_user_counter(5305715501.pw.digitalpath.net) - decrement inUse counter Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 208.53.92.4:5060;branch=z9hG4bK47dfe7eb To: sip:5308997787@216.229.127.55;tag=1c1635606641 From: "DIGITAL PATH" ;tag=as53a24f4b Call-ID: 314856d01c16022c0bfb02fd2817548e@208.53.92.4 CSeq: 103 BYE Contact: sip:99925@66.53.126.36 Record-Route: Supported: em, timer, replaces, path Allow: REGISTER, OPTIONS, INVITE, ACK, CANCEL, BYE, NOTIFY, PRACK, REFER, INFO, SUBSCRIBE, UPDATE Content-Length: 0 11 headers, 0 lines Message is BYE Destroying call '314856d01c16022c0bfb02fd2817548e@208.53.92.4'