root@satish-desktop:/usr/local/src/asterisk/asterisk-1.8.3.3# cat /var/log/asterisk/full [Apr 28 16:11:09] NOTICE[8569] cdr.c: CDR simple logging enabled. [Apr 28 16:11:09] NOTICE[8569] loader.c: 202 modules will be loaded. [Apr 28 16:11:09] NOTICE[8569] res_odbc.c: res_odbc loaded. [Apr 28 16:11:09] NOTICE[8569] res_smdi.c: No SMDI interfaces are available to listen on, not starting SMDI listener. [Apr 28 16:11:09] NOTICE[8569] config.c: Registered Config Engine odbc [Apr 28 16:11:10] NOTICE[8569] chan_iax2.c: IAX/Registry astdb host:port invalid - '172.30.1.47:4569' [Apr 28 16:11:10] VERBOSE[8569] chan_sip.c: SIP channel loading... [Apr 28 16:11:10] WARNING[8569] chan_dahdi.c: Ignoring any changes to 'userbase' (on reload) at line 23. [Apr 28 16:11:10] WARNING[8569] chan_dahdi.c: Ignoring any changes to 'vmsecret' (on reload) at line 31. [Apr 28 16:11:10] WARNING[8569] chan_dahdi.c: Ignoring any changes to 'hassip' (on reload) at line 35. [Apr 28 16:11:10] WARNING[8569] chan_dahdi.c: Ignoring any changes to 'hasiax' (on reload) at line 39. [Apr 28 16:11:10] WARNING[8569] chan_dahdi.c: Ignoring any changes to 'hasmanager' (on reload) at line 47. [Apr 28 16:11:10] NOTICE[8569] chan_skinny.c: Configuring skinny from skinny.conf [Apr 28 16:11:10] NOTICE[8597] chan_sip.c: Peer '7207' is now Reachable. (9ms / 2000ms) [Apr 28 16:11:10] NOTICE[8569] cel_tds.c: cel_tds has no global category, nothing to configure. [Apr 28 16:11:10] WARNING[8569] cel_tds.c: cel_tds module had config problems; declining load [Apr 28 16:11:10] VERBOSE[8569] res_clialiases.c: == Aliased CLI command 'hangup request' to 'channel request hangup' [Apr 28 16:11:10] VERBOSE[8569] res_clialiases.c: == Aliased CLI command 'originate' to 'channel originate' [Apr 28 16:11:10] VERBOSE[8569] res_clialiases.c: == Aliased CLI command 'help' to 'core show help' [Apr 28 16:11:10] VERBOSE[8569] res_clialiases.c: == Aliased CLI command 'pri intense debug span' to 'pri set debug 2 span' [Apr 28 16:11:10] VERBOSE[8569] res_clialiases.c: == Aliased CLI command 'reload' to 'module reload' [Apr 28 16:11:10] NOTICE[8569] pbx_ael.c: Starting AEL load process. [Apr 28 16:11:10] NOTICE[8569] pbx_ael.c: AEL load process: parsed config file name '/etc/asterisk/extensions.ael'. [Apr 28 16:11:10] NOTICE[8569] pbx_ael.c: AEL load process: checked config file name '/etc/asterisk/extensions.ael'. [Apr 28 16:11:10] NOTICE[8569] pbx_ael.c: AEL load process: compiled config file name '/etc/asterisk/extensions.ael'. [Apr 28 16:11:10] NOTICE[8569] pbx_ael.c: AEL load process: merged config file name '/etc/asterisk/extensions.ael'. [Apr 28 16:11:10] NOTICE[8569] pbx_ael.c: AEL load process: verified config file name '/etc/asterisk/extensions.ael'. [Apr 28 16:11:10] NOTICE[8569] app_queue.c: Queue members successfully reloaded from database. [Apr 28 16:11:27] VERBOSE[8597] chan_sip.c: <--- SIP read from UDP:172.30.254.222:52602 ---> <-------------> [Apr 28 16:11:33] VERBOSE[8597] chan_sip.c: <--- SIP read from UDP:172.30.254.222:52602 ---> INVITE sip:7203@172.30.245.208 SIP/2.0 Via: SIP/2.0/UDP 172.30.254.222:52602;branch=z9hG4bK-d87543-8145ad3a82540025-1--d87543-;rport Max-Forwards: 70 Contact: To: "7203" From: "Satish Patel";tag=3c08f01a Call-ID: NjU3NmRmNWZlN2IwNjRiNDdmODFiMWM0NTU2ZGRjN2U. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: X-Lite release 1011s stamp 41150 Content-Length: 241 v=0 o=- 9 2 IN IP4 172.30.254.222 s=CounterPath X-Lite 3.0 c=IN IP4 172.30.254.222 t=0 0 m=audio 43508 RTP/AVP 0 8 101 a=alt:1 1 : 18GKzuEB 14YLqMBY 172.30.254.222 43508 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv <-------------> [Apr 28 16:11:33] VERBOSE[8597] chan_sip.c: --- (12 headers 10 lines) --- [Apr 28 16:11:33] VERBOSE[8597] chan_sip.c: Sending to 172.30.254.222:52602 (no NAT) [Apr 28 16:11:33] VERBOSE[8597] chan_sip.c: Using INVITE request as basis request - NjU3NmRmNWZlN2IwNjRiNDdmODFiMWM0NTU2ZGRjN2U. [Apr 28 16:11:33] VERBOSE[8597] chan_sip.c: Found peer '7207' for '7207' from 172.30.254.222:52602 [Apr 28 16:11:33] VERBOSE[8597] netsock2.c: == Using SIP RTP CoS mark 5 [Apr 28 16:11:33] VERBOSE[8597] chan_sip.c: Found RTP audio format 0 [Apr 28 16:11:33] VERBOSE[8597] chan_sip.c: Found RTP audio format 8 [Apr 28 16:11:33] VERBOSE[8597] chan_sip.c: Found RTP audio format 101 [Apr 28 16:11:33] VERBOSE[8597] chan_sip.c: Found audio description format telephone-event for ID 101 [Apr 28 16:11:33] VERBOSE[8597] chan_sip.c: Capabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) [Apr 28 16:11:33] VERBOSE[8597] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Apr 28 16:11:33] VERBOSE[8597] chan_sip.c: Peer audio RTP is at port 172.30.254.222:43508 [Apr 28 16:11:33] VERBOSE[8597] chan_sip.c: Looking for 7203 in from-sip (domain 172.30.245.208) [Apr 28 16:11:33] VERBOSE[8597] chan_sip.c: list_route: hop: [Apr 28 16:11:33] VERBOSE[8597] chan_sip.c: <--- Transmitting (no NAT) to 172.30.254.222:52602 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.30.254.222:52602;branch=z9hG4bK-d87543-8145ad3a82540025-1--d87543-;received=172.30.254.222;rport=52602 From: "Satish Patel";tag=3c08f01a To: "7203" Call-ID: NjU3NmRmNWZlN2IwNjRiNDdmODFiMWM0NTU2ZGRjN2U. CSeq: 1 INVITE Server: Asterisk PBX 1.8.3.3 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> [Apr 28 16:11:33] VERBOSE[8614] pbx.c: -- Executing [7203@from-sip:1] Macro("SIP/7207-00000000", "stdexten,7203,sip/7203") in new stack [Apr 28 16:11:33] VERBOSE[8614] pbx.c: -- Executing [s@macro-stdexten:1] Dial("SIP/7207-00000000", "sip/7203,10,t") in new stack [Apr 28 16:11:33] VERBOSE[8614] netsock2.c: == Using SIP RTP CoS mark 5 [Apr 28 16:11:33] WARNING[8614] acl.c: Cannot connect [Apr 28 16:11:33] VERBOSE[8614] chan_sip.c: Audio is at 5060 [Apr 28 16:11:33] VERBOSE[8614] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Apr 28 16:11:33] VERBOSE[8614] chan_sip.c: Adding codec 0x2 (gsm) to SDP [Apr 28 16:11:33] VERBOSE[8614] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Apr 28 16:11:33] VERBOSE[8614] chan_sip.c: Adding codec 0x800000000000 (testlaw) to SDP [Apr 28 16:11:33] VERBOSE[8614] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Apr 28 16:11:33] VERBOSE[8614] chan_sip.c: Reliably Transmitting (no NAT) to 0.0.28.35:5060: INVITE sip:7203 SIP/2.0 Via: SIP/2.0/UDP [::1]:5060;branch=z9hG4bK5a052126 Max-Forwards: 70 From: "iPhone" ;tag=as77a0b457 To: Contact: Call-ID: 777ccd9315667f155fbcb3d0366598a5@[::1]:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.3.3 Date: Thu, 28 Apr 2011 20:11:33 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 263 v=0 o=root 713869563 713869563 IN IP6 ::1 s=Asterisk PBX 1.8.3.3 c=IN IP6 ::1 t=0 0 m=audio 19280 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Apr 28 16:11:33] WARNING[8614] chan_sip.c: sip_xmit of 0x8cffa30 (len 762) to 0.0.28.35:5060 returned -1: Invalid argument [Apr 28 16:11:33] VERBOSE[8614] app_dial.c: -- Called 7203 [Apr 28 16:11:33] VERBOSE[8597] chan_sip.c: Retransmitting #1 (no NAT) to 0.0.28.35:5060: INVITE sip:7203 SIP/2.0 Via: SIP/2.0/UDP [::1]:5060;branch=z9hG4bK5a052126 Max-Forwards: 70 From: "iPhone" ;tag=as77a0b457 To: Contact: Call-ID: 777ccd9315667f155fbcb3d0366598a5@[::1]:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.3.3 Date: Thu, 28 Apr 2011 20:11:33 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 263 v=0 o=root 713869563 713869563 IN IP6 ::1 s=Asterisk PBX 1.8.3.3 c=IN IP6 ::1 t=0 0 m=audio 19280 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Apr 28 16:11:33] WARNING[8597] chan_sip.c: sip_xmit of 0x8cffa30 (len 762) to 0.0.28.35:5060 returned -1: Invalid argument [Apr 28 16:11:34] VERBOSE[8597] chan_sip.c: Retransmitting #2 (no NAT) to 0.0.28.35:5060: INVITE sip:7203 SIP/2.0 Via: SIP/2.0/UDP [::1]:5060;branch=z9hG4bK5a052126 Max-Forwards: 70 From: "iPhone" ;tag=as77a0b457 To: Contact: Call-ID: 777ccd9315667f155fbcb3d0366598a5@[::1]:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.3.3 Date: Thu, 28 Apr 2011 20:11:33 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 263 v=0 o=root 713869563 713869563 IN IP6 ::1 s=Asterisk PBX 1.8.3.3 c=IN IP6 ::1 t=0 0 m=audio 19280 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Apr 28 16:11:34] WARNING[8597] chan_sip.c: sip_xmit of 0x8cffa30 (len 762) to 0.0.28.35:5060 returned -1: Invalid argument [Apr 28 16:11:36] VERBOSE[8597] chan_sip.c: Retransmitting #3 (no NAT) to 0.0.28.35:5060: INVITE sip:7203 SIP/2.0 Via: SIP/2.0/UDP [::1]:5060;branch=z9hG4bK5a052126 Max-Forwards: 70 From: "iPhone" ;tag=as77a0b457 To: Contact: Call-ID: 777ccd9315667f155fbcb3d0366598a5@[::1]:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.3.3 Date: Thu, 28 Apr 2011 20:11:33 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 263 v=0 o=root 713869563 713869563 IN IP6 ::1 s=Asterisk PBX 1.8.3.3 c=IN IP6 ::1 t=0 0 m=audio 19280 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Apr 28 16:11:36] WARNING[8597] chan_sip.c: sip_xmit of 0x8cffa30 (len 762) to 0.0.28.35:5060 returned -1: Invalid argument [Apr 28 16:11:40] VERBOSE[8597] chan_sip.c: Retransmitting #4 (no NAT) to 0.0.28.35:5060: INVITE sip:7203 SIP/2.0 Via: SIP/2.0/UDP [::1]:5060;branch=z9hG4bK5a052126 Max-Forwards: 70 From: "iPhone" ;tag=as77a0b457 To: Contact: Call-ID: 777ccd9315667f155fbcb3d0366598a5@[::1]:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.3.3 Date: Thu, 28 Apr 2011 20:11:33 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 263 v=0 o=root 713869563 713869563 IN IP6 ::1 s=Asterisk PBX 1.8.3.3 c=IN IP6 ::1 t=0 0 m=audio 19280 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Apr 28 16:11:40] WARNING[8597] chan_sip.c: sip_xmit of 0x8cffa30 (len 762) to 0.0.28.35:5060 returned -1: Invalid argument [Apr 28 16:11:40] VERBOSE[8613] asterisk.c: -- Remote UNIX connection disconnected [Apr 28 16:11:43] VERBOSE[8614] app_dial.c: -- Nobody picked up in 10000 ms [Apr 28 16:11:43] VERBOSE[8614] chan_sip.c: Scheduling destruction of SIP dialog '777ccd9315667f155fbcb3d0366598a5@[::1]:5060' in 32000 ms (Method: INVITE) [Apr 28 16:11:43] VERBOSE[8614] pbx.c: -- Executing [s@macro-stdexten:2] Goto("SIP/7207-00000000", "s-NOANSWER,1") in new stack [Apr 28 16:11:43] VERBOSE[8614] pbx.c: -- Goto (macro-stdexten,s-NOANSWER,1) [Apr 28 16:11:43] VERBOSE[8614] pbx.c: -- Executing [s-NOANSWER@macro-stdexten:1] VoiceMail("SIP/7207-00000000", "7203,u") in new stack [Apr 28 16:11:43] VERBOSE[8614] chan_sip.c: Audio is at 5060 [Apr 28 16:11:43] VERBOSE[8614] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Apr 28 16:11:43] VERBOSE[8614] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Apr 28 16:11:43] VERBOSE[8614] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Apr 28 16:11:43] VERBOSE[8614] chan_sip.c: <--- Reliably Transmitting (no NAT) to 172.30.254.222:52602 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.30.254.222:52602;branch=z9hG4bK-d87543-8145ad3a82540025-1--d87543-;received=172.30.254.222;rport=52602 From: "Satish Patel";tag=3c08f01a To: "7203";tag=as082346fb Call-ID: NjU3NmRmNWZlN2IwNjRiNDdmODFiMWM0NTU2ZGRjN2U. CSeq: 1 INVITE Server: Asterisk PBX 1.8.3.3 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 264 v=0 o=root 1452095154 1452095154 IN IP4 172.30.245.208 s=Asterisk PBX 1.8.3.3 c=IN IP4 172.30.245.208 t=0 0 m=audio 13830 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <------------> [Apr 28 16:11:43] WARNING[8614] app_voicemail.c: No entry in voicemail config file for '7203' [Apr 28 16:11:43] VERBOSE[8614] pbx.c: -- Executing [s-NOANSWER@macro-stdexten:2] Goto("SIP/7207-00000000", "default,s,1") in new stack [Apr 28 16:11:43] VERBOSE[8614] pbx.c: -- Goto (default,s,1) [Apr 28 16:11:43] VERBOSE[8614] app_macro.c: == Channel 'SIP/7207-00000000' jumping out of macro 'stdexten' [Apr 28 16:11:43] VERBOSE[8614] pbx.c: -- Sent into invalid extension 's' in context 'default' on SIP/7207-00000000 [Apr 28 16:11:43] VERBOSE[8614] pbx.c: -- Executing [i@default:1] Playback("SIP/7207-00000000", "invalid") in new stack [Apr 28 16:11:43] VERBOSE[8614] file.c: -- Playing 'invalid.ulaw' (language 'en') [Apr 28 16:11:43] VERBOSE[8597] chan_sip.c: Retransmitting #1 (no NAT) to 172.30.254.222:52602: SIP/2.0 200 OK Via: SIP/2.0/UDP 172.30.254.222:52602;branch=z9hG4bK-d87543-8145ad3a82540025-1--d87543-;received=172.30.254.222;rport=52602 From: "Satish Patel";tag=3c08f01a To: "7203";tag=as082346fb Call-ID: NjU3NmRmNWZlN2IwNjRiNDdmODFiMWM0NTU2ZGRjN2U. CSeq: 1 INVITE Server: Asterisk PBX 1.8.3.3 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 264 v=0 o=root 1452095154 1452095154 IN IP4 172.30.245.208 s=Asterisk PBX 1.8.3.3 c=IN IP4 172.30.245.208 t=0 0 m=audio 13830 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Apr 28 16:11:43] VERBOSE[8597] chan_sip.c: <--- SIP read from UDP:172.30.254.222:52602 ---> ACK sip:7203@172.30.245.208:5060 SIP/2.0 Via: SIP/2.0/UDP 172.30.254.222:52602;branch=z9hG4bK-d87543-39550a03fb549515-1--d87543-;rport Max-Forwards: 70 Contact: To: "7203";tag=as082346fb From: "Satish Patel";tag=3c08f01a Call-ID: NjU3NmRmNWZlN2IwNjRiNDdmODFiMWM0NTU2ZGRjN2U. CSeq: 1 ACK User-Agent: X-Lite release 1011s stamp 41150 Content-Length: 0 <-------------> [Apr 28 16:11:43] VERBOSE[8597] chan_sip.c: --- (10 headers 0 lines) --- [Apr 28 16:11:43] VERBOSE[8597] chan_sip.c: <--- SIP read from UDP:172.30.254.222:52602 ---> ACK sip:7203@172.30.245.208:5060 SIP/2.0 Via: SIP/2.0/UDP 172.30.254.222:52602;branch=z9hG4bK-d87543-39550a03fb549515-1--d87543-;rport Max-Forwards: 70 Contact: To: "7203";tag=as082346fb From: "Satish Patel";tag=3c08f01a Call-ID: NjU3NmRmNWZlN2IwNjRiNDdmODFiMWM0NTU2ZGRjN2U. CSeq: 1 ACK User-Agent: X-Lite release 1011s stamp 41150 Content-Length: 0 <-------------> [Apr 28 16:11:43] VERBOSE[8597] chan_sip.c: --- (10 headers 0 lines) --- [Apr 28 16:11:47] VERBOSE[8614] pbx.c: -- Executing [i@default:2] Hangup("SIP/7207-00000000", "") in new stack [Apr 28 16:11:47] VERBOSE[8614] pbx.c: == Spawn extension (default, i, 2) exited non-zero on 'SIP/7207-00000000' [Apr 28 16:11:47] VERBOSE[8614] chan_sip.c: Scheduling destruction of SIP dialog 'NjU3NmRmNWZlN2IwNjRiNDdmODFiMWM0NTU2ZGRjN2U.' in 6400 ms (Method: ACK) [Apr 28 16:11:47] VERBOSE[8614] chan_sip.c: set_destination: Parsing for address/port to send to [Apr 28 16:11:47] VERBOSE[8614] chan_sip.c: set_destination: set destination to 172.30.254.222:52602 [Apr 28 16:11:47] VERBOSE[8614] chan_sip.c: Reliably Transmitting (no NAT) to 172.30.254.222:52602: BYE sip:7207@172.30.254.222:52602 SIP/2.0 Via: SIP/2.0/UDP 172.30.245.208:5060;branch=z9hG4bK523f206d;rport Max-Forwards: 70 From: "7203";tag=as082346fb To: "Satish Patel";tag=3c08f01a Call-ID: NjU3NmRmNWZlN2IwNjRiNDdmODFiMWM0NTU2ZGRjN2U. CSeq: 102 BYE User-Agent: Asterisk PBX 1.8.3.3 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [Apr 28 16:11:47] VERBOSE[8597] chan_sip.c: Retransmitting #1 (no NAT) to 172.30.254.222:52602: BYE sip:7207@172.30.254.222:52602 SIP/2.0 Via: SIP/2.0/UDP 172.30.245.208:5060;branch=z9hG4bK523f206d;rport Max-Forwards: 70 From: "7203";tag=as082346fb To: "Satish Patel";tag=3c08f01a Call-ID: NjU3NmRmNWZlN2IwNjRiNDdmODFiMWM0NTU2ZGRjN2U. CSeq: 102 BYE User-Agent: Asterisk PBX 1.8.3.3 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [Apr 28 16:11:47] VERBOSE[8597] chan_sip.c: <--- SIP read from UDP:172.30.254.222:52602 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.30.245.208:5060;branch=z9hG4bK523f206d;rport=5060 Contact: To: "Satish Patel";tag=3c08f01a From: "7203";tag=as082346fb Call-ID: NjU3NmRmNWZlN2IwNjRiNDdmODFiMWM0NTU2ZGRjN2U. CSeq: 102 BYE User-Agent: X-Lite release 1011s stamp 41150 Content-Length: 0 <-------------> [Apr 28 16:11:47] VERBOSE[8597] chan_sip.c: --- (9 headers 0 lines) --- [Apr 28 16:11:47] VERBOSE[8597] chan_sip.c: SIP Response message for INCOMING dialog BYE arrived [Apr 28 16:11:47] VERBOSE[8597] chan_sip.c: Really destroying SIP dialog 'NjU3NmRmNWZlN2IwNjRiNDdmODFiMWM0NTU2ZGRjN2U.' Method: ACK [Apr 28 16:11:47] VERBOSE[8597] chan_sip.c: <--- SIP read from UDP:172.30.254.222:52602 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.30.245.208:5060;branch=z9hG4bK523f206d;rport=5060 Contact: To: "Satish Patel";tag=3c08f01a From: "7203";tag=as082346fb Call-ID: NjU3NmRmNWZlN2IwNjRiNDdmODFiMWM0NTU2ZGRjN2U. CSeq: 102 BYE User-Agent: X-Lite release 1011s stamp 41150 Content-Length: 0 <-------------> [Apr 28 16:11:47] VERBOSE[8597] chan_sip.c: --- (9 headers 0 lines) --- [Apr 28 16:11:48] VERBOSE[8597] chan_sip.c: Retransmitting #5 (no NAT) to 0.0.28.35:5060: INVITE sip:7203 SIP/2.0 Via: SIP/2.0/UDP [::1]:5060;branch=z9hG4bK5a052126 Max-Forwards: 70 From: "iPhone" ;tag=as77a0b457 To: Contact: Call-ID: 777ccd9315667f155fbcb3d0366598a5@[::1]:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.3.3 Date: Thu, 28 Apr 2011 20:11:33 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 263 v=0 o=root 713869563 713869563 IN IP6 ::1 s=Asterisk PBX 1.8.3.3 c=IN IP6 ::1 t=0 0 m=audio 19280 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Apr 28 16:11:48] WARNING[8597] chan_sip.c: sip_xmit of 0x8cffa30 (len 762) to 0.0.28.35:5060 returned -1: Invalid argument root@satish-desktop:/usr/local/src/asterisk/asterisk-1.8.3.3# cat /var/log/asterisk/full [Apr 28 16:11:09] NOTICE[8569] cdr.c: CDR simple logging enabled. [Apr 28 16:11:09] NOTICE[8569] loader.c: 202 modules will be loaded. [Apr 28 16:11:09] NOTICE[8569] res_odbc.c: res_odbc loaded. [Apr 28 16:11:09] NOTICE[8569] res_smdi.c: No SMDI interfaces are available to listen on, not starting SMDI listener. [Apr 28 16:11:09] NOTICE[8569] config.c: Registered Config Engine odbc [Apr 28 16:11:10] NOTICE[8569] chan_iax2.c: IAX/Registry astdb host:port invalid - '172.30.1.47:4569' [Apr 28 16:11:10] VERBOSE[8569] chan_sip.c: SIP channel loading... [Apr 28 16:11:10] WARNING[8569] chan_dahdi.c: Ignoring any changes to 'userbase' (on reload) at line 23. [Apr 28 16:11:10] WARNING[8569] chan_dahdi.c: Ignoring any changes to 'vmsecret' (on reload) at line 31. [Apr 28 16:11:10] WARNING[8569] chan_dahdi.c: Ignoring any changes to 'hassip' (on reload) at line 35. [Apr 28 16:11:10] WARNING[8569] chan_dahdi.c: Ignoring any changes to 'hasiax' (on reload) at line 39. [Apr 28 16:11:10] WARNING[8569] chan_dahdi.c: Ignoring any changes to 'hasmanager' (on reload) at line 47. [Apr 28 16:11:10] NOTICE[8569] chan_skinny.c: Configuring skinny from skinny.conf [Apr 28 16:11:10] NOTICE[8597] chan_sip.c: Peer '7207' is now Reachable. (9ms / 2000ms) [Apr 28 16:11:10] NOTICE[8569] cel_tds.c: cel_tds has no global category, nothing to configure. [Apr 28 16:11:10] WARNING[8569] cel_tds.c: cel_tds module had config problems; declining load [Apr 28 16:11:10] VERBOSE[8569] res_clialiases.c: == Aliased CLI command 'hangup request' to 'channel request hangup' [Apr 28 16:11:10] VERBOSE[8569] res_clialiases.c: == Aliased CLI command 'originate' to 'channel originate' [Apr 28 16:11:10] VERBOSE[8569] res_clialiases.c: == Aliased CLI command 'help' to 'core show help' [Apr 28 16:11:10] VERBOSE[8569] res_clialiases.c: == Aliased CLI command 'pri intense debug span' to 'pri set debug 2 span' [Apr 28 16:11:10] VERBOSE[8569] res_clialiases.c: == Aliased CLI command 'reload' to 'module reload' [Apr 28 16:11:10] NOTICE[8569] pbx_ael.c: Starting AEL load process. [Apr 28 16:11:10] NOTICE[8569] pbx_ael.c: AEL load process: parsed config file name '/etc/asterisk/extensions.ael'. [Apr 28 16:11:10] NOTICE[8569] pbx_ael.c: AEL load process: checked config file name '/etc/asterisk/extensions.ael'. [Apr 28 16:11:10] NOTICE[8569] pbx_ael.c: AEL load process: compiled config file name '/etc/asterisk/extensions.ael'. [Apr 28 16:11:10] NOTICE[8569] pbx_ael.c: AEL load process: merged config file name '/etc/asterisk/extensions.ael'. [Apr 28 16:11:10] NOTICE[8569] pbx_ael.c: AEL load process: verified config file name '/etc/asterisk/extensions.ael'. [Apr 28 16:11:10] NOTICE[8569] app_queue.c: Queue members successfully reloaded from database. [Apr 28 16:11:27] VERBOSE[8597] chan_sip.c: <--- SIP read from UDP:172.30.254.222:52602 ---> <-------------> [Apr 28 16:11:33] VERBOSE[8597] chan_sip.c: <--- SIP read from UDP:172.30.254.222:52602 ---> INVITE sip:7203@172.30.245.208 SIP/2.0 Via: SIP/2.0/UDP 172.30.254.222:52602;branch=z9hG4bK-d87543-8145ad3a82540025-1--d87543-;rport Max-Forwards: 70 Contact: To: "7203" From: "Satish Patel";tag=3c08f01a Call-ID: NjU3NmRmNWZlN2IwNjRiNDdmODFiMWM0NTU2ZGRjN2U. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: X-Lite release 1011s stamp 41150 Content-Length: 241 v=0 o=- 9 2 IN IP4 172.30.254.222 s=CounterPath X-Lite 3.0 c=IN IP4 172.30.254.222 t=0 0 m=audio 43508 RTP/AVP 0 8 101 a=alt:1 1 : 18GKzuEB 14YLqMBY 172.30.254.222 43508 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv <-------------> [Apr 28 16:11:33] VERBOSE[8597] chan_sip.c: --- (12 headers 10 lines) --- [Apr 28 16:11:33] VERBOSE[8597] chan_sip.c: Sending to 172.30.254.222:52602 (no NAT) [Apr 28 16:11:33] VERBOSE[8597] chan_sip.c: Using INVITE request as basis request - NjU3NmRmNWZlN2IwNjRiNDdmODFiMWM0NTU2ZGRjN2U. [Apr 28 16:11:33] VERBOSE[8597] chan_sip.c: Found peer '7207' for '7207' from 172.30.254.222:52602 [Apr 28 16:11:33] VERBOSE[8597] netsock2.c: == Using SIP RTP CoS mark 5 [Apr 28 16:11:33] VERBOSE[8597] chan_sip.c: Found RTP audio format 0 [Apr 28 16:11:33] VERBOSE[8597] chan_sip.c: Found RTP audio format 8 [Apr 28 16:11:33] VERBOSE[8597] chan_sip.c: Found RTP audio format 101 [Apr 28 16:11:33] VERBOSE[8597] chan_sip.c: Found audio description format telephone-event for ID 101 [Apr 28 16:11:33] VERBOSE[8597] chan_sip.c: Capabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) [Apr 28 16:11:33] VERBOSE[8597] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Apr 28 16:11:33] VERBOSE[8597] chan_sip.c: Peer audio RTP is at port 172.30.254.222:43508 [Apr 28 16:11:33] VERBOSE[8597] chan_sip.c: Looking for 7203 in from-sip (domain 172.30.245.208) [Apr 28 16:11:33] VERBOSE[8597] chan_sip.c: list_route: hop: [Apr 28 16:11:33] VERBOSE[8597] chan_sip.c: <--- Transmitting (no NAT) to 172.30.254.222:52602 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.30.254.222:52602;branch=z9hG4bK-d87543-8145ad3a82540025-1--d87543-;received=172.30.254.222;rport=52602 From: "Satish Patel";tag=3c08f01a To: "7203" Call-ID: NjU3NmRmNWZlN2IwNjRiNDdmODFiMWM0NTU2ZGRjN2U. CSeq: 1 INVITE Server: Asterisk PBX 1.8.3.3 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> [Apr 28 16:11:33] VERBOSE[8614] pbx.c: -- Executing [7203@from-sip:1] Macro("SIP/7207-00000000", "stdexten,7203,sip/7203") in new stack [Apr 28 16:11:33] VERBOSE[8614] pbx.c: -- Executing [s@macro-stdexten:1] Dial("SIP/7207-00000000", "sip/7203,10,t") in new stack [Apr 28 16:11:33] VERBOSE[8614] netsock2.c: == Using SIP RTP CoS mark 5 [Apr 28 16:11:33] WARNING[8614] acl.c: Cannot connect [Apr 28 16:11:33] VERBOSE[8614] chan_sip.c: Audio is at 5060 [Apr 28 16:11:33] VERBOSE[8614] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Apr 28 16:11:33] VERBOSE[8614] chan_sip.c: Adding codec 0x2 (gsm) to SDP [Apr 28 16:11:33] VERBOSE[8614] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Apr 28 16:11:33] VERBOSE[8614] chan_sip.c: Adding codec 0x800000000000 (testlaw) to SDP [Apr 28 16:11:33] VERBOSE[8614] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Apr 28 16:11:33] VERBOSE[8614] chan_sip.c: Reliably Transmitting (no NAT) to 0.0.28.35:5060: INVITE sip:7203 SIP/2.0 Via: SIP/2.0/UDP [::1]:5060;branch=z9hG4bK5a052126 Max-Forwards: 70 From: "iPhone" ;tag=as77a0b457 To: Contact: Call-ID: 777ccd9315667f155fbcb3d0366598a5@[::1]:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.3.3 Date: Thu, 28 Apr 2011 20:11:33 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 263 v=0 o=root 713869563 713869563 IN IP6 ::1 s=Asterisk PBX 1.8.3.3 c=IN IP6 ::1 t=0 0 m=audio 19280 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Apr 28 16:11:33] WARNING[8614] chan_sip.c: sip_xmit of 0x8cffa30 (len 762) to 0.0.28.35:5060 returned -1: Invalid argument [Apr 28 16:11:33] VERBOSE[8614] app_dial.c: -- Called 7203 [Apr 28 16:11:33] VERBOSE[8597] chan_sip.c: Retransmitting #1 (no NAT) to 0.0.28.35:5060: INVITE sip:7203 SIP/2.0 Via: SIP/2.0/UDP [::1]:5060;branch=z9hG4bK5a052126 Max-Forwards: 70 From: "iPhone" ;tag=as77a0b457 To: Contact: Call-ID: 777ccd9315667f155fbcb3d0366598a5@[::1]:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.3.3 Date: Thu, 28 Apr 2011 20:11:33 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 263 v=0 o=root 713869563 713869563 IN IP6 ::1 s=Asterisk PBX 1.8.3.3 c=IN IP6 ::1 t=0 0 m=audio 19280 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Apr 28 16:11:33] WARNING[8597] chan_sip.c: sip_xmit of 0x8cffa30 (len 762) to 0.0.28.35:5060 returned -1: Invalid argument [Apr 28 16:11:34] VERBOSE[8597] chan_sip.c: Retransmitting #2 (no NAT) to 0.0.28.35:5060: INVITE sip:7203 SIP/2.0 Via: SIP/2.0/UDP [::1]:5060;branch=z9hG4bK5a052126 Max-Forwards: 70 From: "iPhone" ;tag=as77a0b457 To: Contact: Call-ID: 777ccd9315667f155fbcb3d0366598a5@[::1]:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.3.3 Date: Thu, 28 Apr 2011 20:11:33 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 263 v=0 o=root 713869563 713869563 IN IP6 ::1 s=Asterisk PBX 1.8.3.3 c=IN IP6 ::1 t=0 0 m=audio 19280 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Apr 28 16:11:34] WARNING[8597] chan_sip.c: sip_xmit of 0x8cffa30 (len 762) to 0.0.28.35:5060 returned -1: Invalid argument [Apr 28 16:11:36] VERBOSE[8597] chan_sip.c: Retransmitting #3 (no NAT) to 0.0.28.35:5060: INVITE sip:7203 SIP/2.0 Via: SIP/2.0/UDP [::1]:5060;branch=z9hG4bK5a052126 Max-Forwards: 70 From: "iPhone" ;tag=as77a0b457 To: Contact: Call-ID: 777ccd9315667f155fbcb3d0366598a5@[::1]:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.3.3 Date: Thu, 28 Apr 2011 20:11:33 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 263 v=0 o=root 713869563 713869563 IN IP6 ::1 s=Asterisk PBX 1.8.3.3 c=IN IP6 ::1 t=0 0 m=audio 19280 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Apr 28 16:11:36] WARNING[8597] chan_sip.c: sip_xmit of 0x8cffa30 (len 762) to 0.0.28.35:5060 returned -1: Invalid argument [Apr 28 16:11:40] VERBOSE[8597] chan_sip.c: Retransmitting #4 (no NAT) to 0.0.28.35:5060: INVITE sip:7203 SIP/2.0 Via: SIP/2.0/UDP [::1]:5060;branch=z9hG4bK5a052126 Max-Forwards: 70 From: "iPhone" ;tag=as77a0b457 To: Contact: Call-ID: 777ccd9315667f155fbcb3d0366598a5@[::1]:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.3.3 Date: Thu, 28 Apr 2011 20:11:33 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 263 v=0 o=root 713869563 713869563 IN IP6 ::1 s=Asterisk PBX 1.8.3.3 c=IN IP6 ::1 t=0 0 m=audio 19280 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Apr 28 16:11:40] WARNING[8597] chan_sip.c: sip_xmit of 0x8cffa30 (len 762) to 0.0.28.35:5060 returned -1: Invalid argument [Apr 28 16:11:40] VERBOSE[8613] asterisk.c: -- Remote UNIX connection disconnected [Apr 28 16:11:43] VERBOSE[8614] app_dial.c: -- Nobody picked up in 10000 ms [Apr 28 16:11:43] VERBOSE[8614] chan_sip.c: Scheduling destruction of SIP dialog '777ccd9315667f155fbcb3d0366598a5@[::1]:5060' in 32000 ms (Method: INVITE) [Apr 28 16:11:43] VERBOSE[8614] pbx.c: -- Executing [s@macro-stdexten:2] Goto("SIP/7207-00000000", "s-NOANSWER,1") in new stack [Apr 28 16:11:43] VERBOSE[8614] pbx.c: -- Goto (macro-stdexten,s-NOANSWER,1) [Apr 28 16:11:43] VERBOSE[8614] pbx.c: -- Executing [s-NOANSWER@macro-stdexten:1] VoiceMail("SIP/7207-00000000", "7203,u") in new stack [Apr 28 16:11:43] VERBOSE[8614] chan_sip.c: Audio is at 5060 [Apr 28 16:11:43] VERBOSE[8614] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Apr 28 16:11:43] VERBOSE[8614] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Apr 28 16:11:43] VERBOSE[8614] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Apr 28 16:11:43] VERBOSE[8614] chan_sip.c: <--- Reliably Transmitting (no NAT) to 172.30.254.222:52602 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.30.254.222:52602;branch=z9hG4bK-d87543-8145ad3a82540025-1--d87543-;received=172.30.254.222;rport=52602 From: "Satish Patel";tag=3c08f01a To: "7203";tag=as082346fb Call-ID: NjU3NmRmNWZlN2IwNjRiNDdmODFiMWM0NTU2ZGRjN2U. CSeq: 1 INVITE Server: Asterisk PBX 1.8.3.3 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 264 v=0 o=root 1452095154 1452095154 IN IP4 172.30.245.208 s=Asterisk PBX 1.8.3.3 c=IN IP4 172.30.245.208 t=0 0 m=audio 13830 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <------------> [Apr 28 16:11:43] WARNING[8614] app_voicemail.c: No entry in voicemail config file for '7203' [Apr 28 16:11:43] VERBOSE[8614] pbx.c: -- Executing [s-NOANSWER@macro-stdexten:2] Goto("SIP/7207-00000000", "default,s,1") in new stack [Apr 28 16:11:43] VERBOSE[8614] pbx.c: -- Goto (default,s,1) [Apr 28 16:11:43] VERBOSE[8614] app_macro.c: == Channel 'SIP/7207-00000000' jumping out of macro 'stdexten' [Apr 28 16:11:43] VERBOSE[8614] pbx.c: -- Sent into invalid extension 's' in context 'default' on SIP/7207-00000000 [Apr 28 16:11:43] VERBOSE[8614] pbx.c: -- Executing [i@default:1] Playback("SIP/7207-00000000", "invalid") in new stack [Apr 28 16:11:43] VERBOSE[8614] file.c: -- Playing 'invalid.ulaw' (language 'en') [Apr 28 16:11:43] VERBOSE[8597] chan_sip.c: Retransmitting #1 (no NAT) to 172.30.254.222:52602: SIP/2.0 200 OK Via: SIP/2.0/UDP 172.30.254.222:52602;branch=z9hG4bK-d87543-8145ad3a82540025-1--d87543-;received=172.30.254.222;rport=52602 From: "Satish Patel";tag=3c08f01a To: "7203";tag=as082346fb Call-ID: NjU3NmRmNWZlN2IwNjRiNDdmODFiMWM0NTU2ZGRjN2U. CSeq: 1 INVITE Server: Asterisk PBX 1.8.3.3 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 264 v=0 o=root 1452095154 1452095154 IN IP4 172.30.245.208 s=Asterisk PBX 1.8.3.3 c=IN IP4 172.30.245.208 t=0 0 m=audio 13830 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Apr 28 16:11:43] VERBOSE[8597] chan_sip.c: <--- SIP read from UDP:172.30.254.222:52602 ---> ACK sip:7203@172.30.245.208:5060 SIP/2.0 Via: SIP/2.0/UDP 172.30.254.222:52602;branch=z9hG4bK-d87543-39550a03fb549515-1--d87543-;rport Max-Forwards: 70 Contact: To: "7203";tag=as082346fb From: "Satish Patel";tag=3c08f01a Call-ID: NjU3NmRmNWZlN2IwNjRiNDdmODFiMWM0NTU2ZGRjN2U. CSeq: 1 ACK User-Agent: X-Lite release 1011s stamp 41150 Content-Length: 0 <-------------> [Apr 28 16:11:43] VERBOSE[8597] chan_sip.c: --- (10 headers 0 lines) --- [Apr 28 16:11:43] VERBOSE[8597] chan_sip.c: <--- SIP read from UDP:172.30.254.222:52602 ---> ACK sip:7203@172.30.245.208:5060 SIP/2.0 Via: SIP/2.0/UDP 172.30.254.222:52602;branch=z9hG4bK-d87543-39550a03fb549515-1--d87543-;rport Max-Forwards: 70 Contact: To: "7203";tag=as082346fb From: "Satish Patel";tag=3c08f01a Call-ID: NjU3NmRmNWZlN2IwNjRiNDdmODFiMWM0NTU2ZGRjN2U. CSeq: 1 ACK User-Agent: X-Lite release 1011s stamp 41150 Content-Length: 0 <-------------> [Apr 28 16:11:43] VERBOSE[8597] chan_sip.c: --- (10 headers 0 lines) --- [Apr 28 16:11:47] VERBOSE[8614] pbx.c: -- Executing [i@default:2] Hangup("SIP/7207-00000000", "") in new stack [Apr 28 16:11:47] VERBOSE[8614] pbx.c: == Spawn extension (default, i, 2) exited non-zero on 'SIP/7207-00000000' [Apr 28 16:11:47] VERBOSE[8614] chan_sip.c: Scheduling destruction of SIP dialog 'NjU3NmRmNWZlN2IwNjRiNDdmODFiMWM0NTU2ZGRjN2U.' in 6400 ms (Method: ACK) [Apr 28 16:11:47] VERBOSE[8614] chan_sip.c: set_destination: Parsing for address/port to send to [Apr 28 16:11:47] VERBOSE[8614] chan_sip.c: set_destination: set destination to 172.30.254.222:52602 [Apr 28 16:11:47] VERBOSE[8614] chan_sip.c: Reliably Transmitting (no NAT) to 172.30.254.222:52602: BYE sip:7207@172.30.254.222:52602 SIP/2.0 Via: SIP/2.0/UDP 172.30.245.208:5060;branch=z9hG4bK523f206d;rport Max-Forwards: 70 From: "7203";tag=as082346fb To: "Satish Patel";tag=3c08f01a Call-ID: NjU3NmRmNWZlN2IwNjRiNDdmODFiMWM0NTU2ZGRjN2U. CSeq: 102 BYE User-Agent: Asterisk PBX 1.8.3.3 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [Apr 28 16:11:47] VERBOSE[8597] chan_sip.c: Retransmitting #1 (no NAT) to 172.30.254.222:52602: BYE sip:7207@172.30.254.222:52602 SIP/2.0 Via: SIP/2.0/UDP 172.30.245.208:5060;branch=z9hG4bK523f206d;rport Max-Forwards: 70 From: "7203";tag=as082346fb To: "Satish Patel";tag=3c08f01a Call-ID: NjU3NmRmNWZlN2IwNjRiNDdmODFiMWM0NTU2ZGRjN2U. CSeq: 102 BYE User-Agent: Asterisk PBX 1.8.3.3 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [Apr 28 16:11:47] VERBOSE[8597] chan_sip.c: <--- SIP read from UDP:172.30.254.222:52602 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.30.245.208:5060;branch=z9hG4bK523f206d;rport=5060 Contact: To: "Satish Patel";tag=3c08f01a From: "7203";tag=as082346fb Call-ID: NjU3NmRmNWZlN2IwNjRiNDdmODFiMWM0NTU2ZGRjN2U. CSeq: 102 BYE User-Agent: X-Lite release 1011s stamp 41150 Content-Length: 0 <-------------> [Apr 28 16:11:47] VERBOSE[8597] chan_sip.c: --- (9 headers 0 lines) --- [Apr 28 16:11:47] VERBOSE[8597] chan_sip.c: SIP Response message for INCOMING dialog BYE arrived [Apr 28 16:11:47] VERBOSE[8597] chan_sip.c: Really destroying SIP dialog 'NjU3NmRmNWZlN2IwNjRiNDdmODFiMWM0NTU2ZGRjN2U.' Method: ACK [Apr 28 16:11:47] VERBOSE[8597] chan_sip.c: <--- SIP read from UDP:172.30.254.222:52602 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.30.245.208:5060;branch=z9hG4bK523f206d;rport=5060 Contact: To: "Satish Patel";tag=3c08f01a From: "7203";tag=as082346fb Call-ID: NjU3NmRmNWZlN2IwNjRiNDdmODFiMWM0NTU2ZGRjN2U. CSeq: 102 BYE User-Agent: X-Lite release 1011s stamp 41150 Content-Length: 0 <-------------> [Apr 28 16:11:47] VERBOSE[8597] chan_sip.c: --- (9 headers 0 lines) --- [Apr 28 16:11:48] VERBOSE[8597] chan_sip.c: Retransmitting #5 (no NAT) to 0.0.28.35:5060: INVITE sip:7203 SIP/2.0 Via: SIP/2.0/UDP [::1]:5060;branch=z9hG4bK5a052126 Max-Forwards: 70 From: "iPhone" ;tag=as77a0b457 To: Contact: Call-ID: 777ccd9315667f155fbcb3d0366598a5@[::1]:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.3.3 Date: Thu, 28 Apr 2011 20:11:33 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 263 v=0 o=root 713869563 713869563 IN IP6 ::1 s=Asterisk PBX 1.8.3.3 c=IN IP6 ::1 t=0 0 m=audio 19280 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Apr 28 16:11:48] WARNING[8597] chan_sip.c: sip_xmit of 0x8cffa30 (len 762) to 0.0.28.35:5060 returned -1: Invalid argument [Apr 28 16:11:57] VERBOSE[8597] chan_sip.c: <--- SIP read from UDP:172.30.254.222:52602 ---> <-------------> [Apr 28 16:12:04] VERBOSE[8597] chan_sip.c: Retransmitting #6 (no NAT) to 0.0.28.35:5060: INVITE sip:7203 SIP/2.0 Via: SIP/2.0/UDP [::1]:5060;branch=z9hG4bK5a052126 Max-Forwards: 70 From: "iPhone" ;tag=as77a0b457 To: Contact: Call-ID: 777ccd9315667f155fbcb3d0366598a5@[::1]:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.3.3 Date: Thu, 28 Apr 2011 20:11:33 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 263 v=0 o=root 713869563 713869563 IN IP6 ::1 s=Asterisk PBX 1.8.3.3 c=IN IP6 ::1 t=0 0 m=audio 19280 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Apr 28 16:12:04] WARNING[8597] chan_sip.c: sip_xmit of 0x8cffa30 (len 762) to 0.0.28.35:5060 returned -1: Invalid argument [Apr 28 16:12:05] WARNING[8597] chan_sip.c: Retransmission timeout reached on transmission 777ccd9315667f155fbcb3d0366598a5@[::1]:5060 for seqno 102 (Critical Request) -- See doc/sip-retransmit.txt. Packet timed out after 32000ms with no response [Apr 28 16:12:05] VERBOSE[8597] chan_sip.c: Really destroying SIP dialog '777ccd9315667f155fbcb3d0366598a5@[::1]:5060' Method: INVITE [Apr 28 16:12:10] VERBOSE[8597] chan_sip.c: Reliably Transmitting (no NAT) to 172.30.254.222:52602: OPTIONS sip:7207@172.30.254.222:52602;rinstance=3f18588797b277fb SIP/2.0 Via: SIP/2.0/UDP 172.30.245.208:5060;branch=z9hG4bK659b0fc5 Max-Forwards: 70 From: "asterisk" ;tag=as3ad9176d To: Contact: Call-ID: 19c9f8f87850cb011ab78b073068d3c6@172.30.245.208:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.3.3 Date: Thu, 28 Apr 2011 20:12:10 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Apr 28 16:12:10] VERBOSE[8597] chan_sip.c: <--- SIP read from UDP:172.30.254.222:52602 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.30.245.208:5060;branch=z9hG4bK659b0fc5 Contact: To: ;tag=1368c90f From: "asterisk";tag=as3ad9176d Call-ID: 19c9f8f87850cb011ab78b073068d3c6@172.30.245.208:5060 CSeq: 102 OPTIONS Accept: application/sdp Accept-Language: en Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: X-Lite release 1011s stamp 41150 Content-Length: 0 <-------------> [Apr 28 16:12:10] VERBOSE[8597] chan_sip.c: --- (12 headers 0 lines) --- [Apr 28 16:12:10] VERBOSE[8597] chan_sip.c: Really destroying SIP dialog '19c9f8f87850cb011ab78b073068d3c6@172.30.245.208:5060' Method: OPTIONS [Apr 28 16:12:27] VERBOSE[8597] chan_sip.c: <--- SIP read from UDP:172.30.254.222:52602 ---> <-------------> [Apr 28 16:12:57] VERBOSE[8597] chan_sip.c: <--- SIP read from UDP:172.30.254.222:52602 ---> <-------------> [Apr 28 16:13:10] VERBOSE[8597] chan_sip.c: Reliably Transmitting (no NAT) to 172.30.254.222:52602: OPTIONS sip:7207@172.30.254.222:52602;rinstance=3f18588797b277fb SIP/2.0 Via: SIP/2.0/UDP 172.30.245.208:5060;branch=z9hG4bK7c4bf879 Max-Forwards: 70 From: "asterisk" ;tag=as6736639b To: Contact: Call-ID: 50f50f3463d9082429a5e9644861aa7b@172.30.245.208:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.3.3 Date: Thu, 28 Apr 2011 20:13:10 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Apr 28 16:13:10] VERBOSE[8597] chan_sip.c: <--- SIP read from UDP:172.30.254.222:52602 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.30.245.208:5060;branch=z9hG4bK7c4bf879 Contact: To: ;tag=692c4e12 From: "asterisk";tag=as6736639b Call-ID: 50f50f3463d9082429a5e9644861aa7b@172.30.245.208:5060 CSeq: 102 OPTIONS Accept: application/sdp Accept-Language: en Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: X-Lite release 1011s stamp 41150 Content-Length: 0 <-------------> [Apr 28 16:13:10] VERBOSE[8597] chan_sip.c: --- (12 headers 0 lines) --- [Apr 28 16:13:10] VERBOSE[8597] chan_sip.c: Really destroying SIP dialog '50f50f3463d9082429a5e9644861aa7b@172.30.245.208:5060' Method: OPTIONS [Apr 28 16:13:27] VERBOSE[8597] chan_sip.c: <--- SIP read from UDP:172.30.254.222:52602 ---> <------------->