*CLI> sip set debug on SIP Debugging enabled *CLI> core set verbose 100 Verbosity was 4 and is now 100 *CLI> <--- 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-d87543-0972fa387054631d-1--d87543-;rport Max-Forwards: 70 Contact: To: "7203" From: "Satish Patel";tag=6b522754 Call-ID: NmJiNDFmYjEyOWQ4YjI4OWE2YTJmOTAzYzJlOTBmMWU. 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=- 1 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 12524 RTP/AVP 0 8 101 a=alt:1 1 : USMEbxQz dAW/vnqm 172.30.254.222 12524 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv <-------------> --- (12 headers 10 lines) --- Sending to 172.30.254.222:52602 (no NAT) Using INVITE request as basis request - NmJiNDFmYjEyOWQ4YjI4OWE2YTJmOTAzYzJlOTBmMWU. Found peer '7207' for '7207' from 172.30.254.222:52602 == Using SIP RTP CoS mark 5 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 Found audio description format telephone-event for ID 101 Capabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 172.30.254.222:12524 Looking for 7203 in from-sip (domain 172.30.245.208) list_route: hop: <--- 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-d87543-0972fa387054631d-1--d87543-;received=172.30.254.222;rport=52602 From: "Satish Patel";tag=6b522754 To: "7203" Call-ID: NmJiNDFmYjEyOWQ4YjI4OWE2YTJmOTAzYzJlOTBmMWU. 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 <------------> -- Executing [7203@from-sip:1] Macro("SIP/7207-00000000", "stdexten,7203,sip/7203") in new stack -- Executing [s@macro-stdexten:1] Dial("SIP/7207-00000000", "sip/7203,10,t") in new stack == Using SIP RTP CoS mark 5 [Apr 28 16:05:56] WARNING[8361]: acl.c:698 ast_ouraddrfor: Cannot connect Audio is at 5060 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x800000000000 (testlaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 0.0.28.35:5060: INVITE sip:7203 SIP/2.0 Via: SIP/2.0/UDP [::1]:5060;branch=z9hG4bK303c4b40 Max-Forwards: 70 From: "iPhone" ;tag=as2f452ace To: Contact: Call-ID: 11ac8ea9063719a6192ea6fc225b58cd@[::1]:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.3.3 Date: Thu, 28 Apr 2011 20:05:56 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 265 v=0 o=root 1586096460 1586096460 IN IP6 ::1 s=Asterisk PBX 1.8.3.3 c=IN IP6 ::1 t=0 0 m=audio 18940 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 16:05:56] WARNING[8361]: chan_sip.c:3240 __sip_xmit: sip_xmit of 0x2888590 (len 764) to 0.0.28.35:5060 returned -1: Invalid argument -- Called 7203 <--- SIP read from UDP:172.30.254.222:52602 ---> <-------------> Retransmitting #1 (no NAT) to 0.0.28.35:5060: INVITE sip:7203 SIP/2.0 Via: SIP/2.0/UDP [::1]:5060;branch=z9hG4bK303c4b40 Max-Forwards: 70 From: "iPhone" ;tag=as2f452ace To: Contact: Call-ID: 11ac8ea9063719a6192ea6fc225b58cd@[::1]:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.3.3 Date: Thu, 28 Apr 2011 20:05:56 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 265 v=0 o=root 1586096460 1586096460 IN IP6 ::1 s=Asterisk PBX 1.8.3.3 c=IN IP6 ::1 t=0 0 m=audio 18940 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 16:05:56] WARNING[8345]: chan_sip.c:3240 __sip_xmit: sip_xmit of 0x2888590 (len 764) to 0.0.28.35:5060 returned -1: Invalid argument Retransmitting #2 (no NAT) to 0.0.28.35:5060: INVITE sip:7203 SIP/2.0 Via: SIP/2.0/UDP [::1]:5060;branch=z9hG4bK303c4b40 Max-Forwards: 70 From: "iPhone" ;tag=as2f452ace To: Contact: Call-ID: 11ac8ea9063719a6192ea6fc225b58cd@[::1]:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.3.3 Date: Thu, 28 Apr 2011 20:05:56 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 265 v=0 o=root 1586096460 1586096460 IN IP6 ::1 s=Asterisk PBX 1.8.3.3 c=IN IP6 ::1 t=0 0 m=audio 18940 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 16:05:57] WARNING[8345]: chan_sip.c:3240 __sip_xmit: sip_xmit of 0x2888590 (len 764) to 0.0.28.35:5060 returned -1: Invalid argument Retransmitting #3 (no NAT) to 0.0.28.35:5060: INVITE sip:7203 SIP/2.0 Via: SIP/2.0/UDP [::1]:5060;branch=z9hG4bK303c4b40 Max-Forwards: 70 From: "iPhone" ;tag=as2f452ace To: Contact: Call-ID: 11ac8ea9063719a6192ea6fc225b58cd@[::1]:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.3.3 Date: Thu, 28 Apr 2011 20:05:56 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 265 v=0 o=root 1586096460 1586096460 IN IP6 ::1 s=Asterisk PBX 1.8.3.3 c=IN IP6 ::1 t=0 0 m=audio 18940 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 16:05:59] WARNING[8345]: chan_sip.c:3240 __sip_xmit: sip_xmit of 0x2888590 (len 764) to 0.0.28.35:5060 returned -1: Invalid argument 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=z9hG4bK67ec11ad Max-Forwards: 70 From: "asterisk" ;tag=as4d28425c To: Contact: Call-ID: 65e61054139f631d41d3a8f51779d438@172.30.245.208:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.3.3 Date: Thu, 28 Apr 2011 20:06:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- <--- 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=z9hG4bK67ec11ad Contact: To: ;tag=3221d447 From: "asterisk";tag=as4d28425c Call-ID: 65e61054139f631d41d3a8f51779d438@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 <-------------> --- (12 headers 0 lines) --- Really destroying SIP dialog '65e61054139f631d41d3a8f51779d438@172.30.245.208:5060' Method: OPTIONS Retransmitting #4 (no NAT) to 0.0.28.35:5060: INVITE sip:7203 SIP/2.0 Via: SIP/2.0/UDP [::1]:5060;branch=z9hG4bK303c4b40 Max-Forwards: 70 From: "iPhone" ;tag=as2f452ace To: Contact: Call-ID: 11ac8ea9063719a6192ea6fc225b58cd@[::1]:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.3.3 Date: Thu, 28 Apr 2011 20:05:56 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 265 v=0 o=root 1586096460 1586096460 IN IP6 ::1 s=Asterisk PBX 1.8.3.3 c=IN IP6 ::1 t=0 0 m=audio 18940 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 16:06:03] WARNING[8345]: chan_sip.c:3240 __sip_xmit: sip_xmit of 0x2888590 (len 764) to 0.0.28.35:5060 returned -1: Invalid argument -- Nobody picked up in 10000 ms Scheduling destruction of SIP dialog '11ac8ea9063719a6192ea6fc225b58cd@[::1]:5060' in 32000 ms (Method: INVITE) -- Executing [s@macro-stdexten:2] Goto("SIP/7207-00000000", "s-NOANSWER,1") in new stack -- Goto (macro-stdexten,s-NOANSWER,1) -- Executing [s-NOANSWER@macro-stdexten:1] VoiceMail("SIP/7207-00000000", "7203,u") in new stack Audio is at 5060 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- 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-d87543-0972fa387054631d-1--d87543-;received=172.30.254.222;rport=52602 From: "Satish Patel";tag=6b522754 To: "7203";tag=as0d7f4e32 Call-ID: NmJiNDFmYjEyOWQ4YjI4OWE2YTJmOTAzYzJlOTBmMWU. 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 1348515958 1348515958 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 18756 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 16:06:06] WARNING[8361]: app_voicemail.c:5535 leave_voicemail: No entry in voicemail config file for '7203' -- Executing [s-NOANSWER@macro-stdexten:2] Goto("SIP/7207-00000000", "default,s,1") in new stack -- Goto (default,s,1) == Channel 'SIP/7207-00000000' jumping out of macro 'stdexten' -- Sent into invalid extension 's' in context 'default' on SIP/7207-00000000 -- Executing [i@default:1] Playback("SIP/7207-00000000", "invalid") in new stack -- Playing 'invalid.ulaw' (language 'en') 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-d87543-0972fa387054631d-1--d87543-;received=172.30.254.222;rport=52602 From: "Satish Patel";tag=6b522754 To: "7203";tag=as0d7f4e32 Call-ID: NmJiNDFmYjEyOWQ4YjI4OWE2YTJmOTAzYzJlOTBmMWU. 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 1348515958 1348515958 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 18756 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 --- <--- 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-b531436ea219290c-1--d87543-;rport Max-Forwards: 70 Contact: To: "7203";tag=as0d7f4e32 From: "Satish Patel";tag=6b522754 Call-ID: NmJiNDFmYjEyOWQ4YjI4OWE2YTJmOTAzYzJlOTBmMWU. CSeq: 1 ACK User-Agent: X-Lite release 1011s stamp 41150 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- 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-b531436ea219290c-1--d87543-;rport Max-Forwards: 70 Contact: To: "7203";tag=as0d7f4e32 From: "Satish Patel";tag=6b522754 Call-ID: NmJiNDFmYjEyOWQ4YjI4OWE2YTJmOTAzYzJlOTBmMWU. CSeq: 1 ACK User-Agent: X-Lite release 1011s stamp 41150 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- -- Executing [i@default:2] Hangup("SIP/7207-00000000", "") in new stack == Spawn extension (default, i, 2) exited non-zero on 'SIP/7207-00000000' Scheduling destruction of SIP dialog 'NmJiNDFmYjEyOWQ4YjI4OWE2YTJmOTAzYzJlOTBmMWU.' in 6400 ms (Method: ACK) set_destination: Parsing for address/port to send to set_destination: set destination to 172.30.254.222:52602 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=z9hG4bK2b2222c3;rport Max-Forwards: 70 From: "7203";tag=as0d7f4e32 To: "Satish Patel";tag=6b522754 Call-ID: NmJiNDFmYjEyOWQ4YjI4OWE2YTJmOTAzYzJlOTBmMWU. CSeq: 102 BYE User-Agent: Asterisk PBX 1.8.3.3 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- 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=z9hG4bK2b2222c3;rport Max-Forwards: 70 From: "7203";tag=as0d7f4e32 To: "Satish Patel";tag=6b522754 Call-ID: NmJiNDFmYjEyOWQ4YjI4OWE2YTJmOTAzYzJlOTBmMWU. CSeq: 102 BYE User-Agent: Asterisk PBX 1.8.3.3 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- <--- 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=z9hG4bK2b2222c3;rport=5060 Contact: To: "Satish Patel";tag=6b522754 From: "7203";tag=as0d7f4e32 Call-ID: NmJiNDFmYjEyOWQ4YjI4OWE2YTJmOTAzYzJlOTBmMWU. CSeq: 102 BYE User-Agent: X-Lite release 1011s stamp 41150 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- SIP Response message for INCOMING dialog BYE arrived Really destroying SIP dialog 'NmJiNDFmYjEyOWQ4YjI4OWE2YTJmOTAzYzJlOTBmMWU.' Method: ACK <--- 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=z9hG4bK2b2222c3;rport=5060 Contact: To: "Satish Patel";tag=6b522754 From: "7203";tag=as0d7f4e32 Call-ID: NmJiNDFmYjEyOWQ4YjI4OWE2YTJmOTAzYzJlOTBmMWU. CSeq: 102 BYE User-Agent: X-Lite release 1011s stamp 41150 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Retransmitting #5 (no NAT) to 0.0.28.35:5060: INVITE sip:7203 SIP/2.0 Via: SIP/2.0/UDP [::1]:5060;branch=z9hG4bK303c4b40 Max-Forwards: 70 From: "iPhone" ;tag=as2f452ace To: Contact: Call-ID: 11ac8ea9063719a6192ea6fc225b58cd@[::1]:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.3.3 Date: Thu, 28 Apr 2011 20:05:56 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 265 v=0 o=root 1586096460 1586096460 IN IP6 ::1 s=Asterisk PBX 1.8.3.3 c=IN IP6 ::1 t=0 0 m=audio 18940 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 16:06:11] WARNING[8345]: chan_sip.c:3240 __sip_xmit: sip_xmit of 0x2888590 (len 764) to 0.0.28.35:5060 returned -1: Invalid argument <--- SIP read from UDP:172.30.254.222:52602 ---> <-------------> Retransmitting #6 (no NAT) to 0.0.28.35:5060: INVITE sip:7203 SIP/2.0 Via: SIP/2.0/UDP [::1]:5060;branch=z9hG4bK303c4b40 Max-Forwards: 70 From: "iPhone" ;tag=as2f452ace To: Contact: Call-ID: 11ac8ea9063719a6192ea6fc225b58cd@[::1]:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.3.3 Date: Thu, 28 Apr 2011 20:05:56 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 265 v=0 o=root 1586096460 1586096460 IN IP6 ::1 s=Asterisk PBX 1.8.3.3 c=IN IP6 ::1 t=0 0 m=audio 18940 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 16:06:27] WARNING[8345]: chan_sip.c:3240 __sip_xmit: sip_xmit of 0x2888590 (len 764) to 0.0.28.35:5060 returned -1: Invalid argument [Apr 28 16:06:28] WARNING[8345]: chan_sip.c:3511 retrans_pkt: Retransmission timeout reached on transmission 11ac8ea9063719a6192ea6fc225b58cd@[::1]:5060 for seqno 102 (Critical Request) -- See doc/sip-retransmit.txt. Packet timed out after 32000ms with no response Really destroying SIP dialog '11ac8ea9063719a6192ea6fc225b58cd@[::1]:5060' Method: INVITE *CLI>