shirley*CLI> core set verbose 15 Verbosity was 100 and is now 15 shirley*CLI> core set debug 15 Core debug was 100 and is now 15 shirley*CLI> module reload logger == Parsing '/etc/asterisk/logger.conf': == Found Asterisk Queue Logger restarted shirley*CLI> logger rotate == Parsing '/etc/asterisk/logger.conf': == Found Asterisk Queue Logger restarted shirley*CLI> sip set debug on SIP Debugging enabled shirley*CLI> iax2 set debug on IAX2 Debugging Enabled shirley*CLI> shirley*CLI> shirley*CLI> shirley*CLI> shirley*CLI> shirley*CLI> shirley*CLI> shirley*CLI> shirley*CLI> <--- SIP read from UDP:172.30.254.48:5060 ---> INVITE sip:7624@172.30.1.47:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 172.30.254.48;branch=z9hG4bK2f2634f4C8756E5D From: "Cambridge Guest" ;tag=DA87FFEE-A33412B9 To: CSeq: 1 INVITE Call-ID: 47e59e82-4cf00900-7c1efcf3@172.30.254.48 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.5.0043 Supported: 100rel,replaces Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 251 v=0 o=- 1302282025 1302282025 IN IP4 172.30.254.48 s=Polycom IP Phone c=IN IP4 172.30.254.48 t=0 0 a=sendrecv m=audio 2230 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 <-------------> --- (14 headers 11 lines) --- Sending to 172.30.254.48:5060 (no NAT) Using INVITE request as basis request - 47e59e82-4cf00900-7c1efcf3@172.30.254.48 Found peer '7527' for '7527' from 172.30.254.48:5060 == Using SIP RTP CoS mark 5 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 101 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format G729 for ID 18 Found audio description format telephone-event for ID 101 Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 172.30.254.48:2230 Looking for 7624 in from-sip (domain 172.30.1.47:5060) list_route: hop: <--- Transmitting (no NAT) to 172.30.254.48:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.30.254.48;branch=z9hG4bK2f2634f4C8756E5D;received=172.30.254.48 From: "Cambridge Guest" ;tag=DA87FFEE-A33412B9 To: Call-ID: 47e59e82-4cf00900-7c1efcf3@172.30.254.48 CSeq: 1 INVITE Server: Asterisk PBX 1.8.3.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> -- Executing [7624@from-sip:1] Macro("SIP/7527-000000d8", "stdexten,7624,SIP/7624") in new stack -- Executing [s@macro-stdexten:1] Dial("SIP/7527-000000d8", "SIP/7624&IAX2/7624,20,t") in new stack == Using SIP RTP CoS mark 5 [Apr 8 12:59:23] WARNING[15249]: acl.c:698 ast_ouraddrfor: Cannot connect Audio is at 5060 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 0.0.29.200:5060: INVITE sip:7624 SIP/2.0 Via: SIP/2.0/UDP 172.30.1.47:5060;branch=z9hG4bK1789e45d Max-Forwards: 70 From: "Cambridge Guest" ;tag=as1f3e09eb To: Contact: Call-ID: 6d8be7981c3219884243c61b4a4de639@172.30.1.47:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.3.2 Date: Fri, 08 Apr 2011 19:59:23 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 257 v=0 o=root 1103328122 1103328122 IN IP4 172.30.1.47 s=Asterisk PBX 1.8.3.2 c=IN IP4 172.30.1.47 t=0 0 m=audio 12384 RTP/AVP 0 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Apr 8 12:59:23] WARNING[15249]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x2f32e00 (len 789) to 0.0.29.200:5060 returned -1: Invalid argument -- Called 7624 Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 00005ms SCall: 04965 DCall: 00000 [0.0.29.200:4569] VERSION : 2 CALLED NUMBER : s CODEC_PREFS : () CALLING NUMBER : 7527 CALLING PRESNTN : 0 CALLING TYPEOFN : 0 CALLING TRANSIT : 0 CALLING NAME : Cambridge Guest LANGUAGE : en FORMAT : 2 FORMAT2 : gsm CAPABILITY : 1795 CAPABILITY2 : unknown ADSICPE : 2 DATE TIME : 2011-04-08 12:59:22 -- Called 7624 Retransmitting #1 (no NAT) to 0.0.29.200:5060: INVITE sip:7624 SIP/2.0 Via: SIP/2.0/UDP 172.30.1.47:5060;branch=z9hG4bK1789e45d Max-Forwards: 70 From: "Cambridge Guest" ;tag=as1f3e09eb To: Contact: Call-ID: 6d8be7981c3219884243c61b4a4de639@172.30.1.47:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.3.2 Date: Fri, 08 Apr 2011 19:59:23 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 257 v=0 o=root 1103328122 1103328122 IN IP4 172.30.1.47 s=Asterisk PBX 1.8.3.2 c=IN IP4 172.30.1.47 t=0 0 m=audio 12384 RTP/AVP 0 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Apr 8 12:59:24] WARNING[13920]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x2f32e00 (len 789) to 0.0.29.200:5060 returned -1: Invalid argument Retransmitting #2 (no NAT) to 0.0.29.200:5060: INVITE sip:7624 SIP/2.0 Via: SIP/2.0/UDP 172.30.1.47:5060;branch=z9hG4bK1789e45d Max-Forwards: 70 From: "Cambridge Guest" ;tag=as1f3e09eb To: Contact: Call-ID: 6d8be7981c3219884243c61b4a4de639@172.30.1.47:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.3.2 Date: Fri, 08 Apr 2011 19:59:23 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 257 v=0 o=root 1103328122 1103328122 IN IP4 172.30.1.47 s=Asterisk PBX 1.8.3.2 c=IN IP4 172.30.1.47 t=0 0 m=audio 12384 RTP/AVP 0 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Apr 8 12:59:25] WARNING[13920]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x2f32e00 (len 789) to 0.0.29.200:5060 returned -1: Invalid argument Tx-Frame Retry[001] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 00005ms SCall: 04965 DCall: 00000 [0.0.29.200:4569] VERSION : 2 CALLED NUMBER : s CODEC_PREFS : () CALLING NUMBER : 7527 CALLING PRESNTN : 0 CALLING TYPEOFN : 0 CALLING TRANSIT : 0 CALLING NAME : Cambridge Guest LANGUAGE : en FORMAT : 2 FORMAT2 : gsm CAPABILITY : 1795 CAPABILITY2 : unknown ADSICPE : 2 DATE TIME : 2011-04-08 12:59:22 Retransmitting #3 (no NAT) to 0.0.29.200:5060: INVITE sip:7624 SIP/2.0 Via: SIP/2.0/UDP 172.30.1.47:5060;branch=z9hG4bK1789e45d Max-Forwards: 70 From: "Cambridge Guest" ;tag=as1f3e09eb To: Contact: Call-ID: 6d8be7981c3219884243c61b4a4de639@172.30.1.47:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.3.2 Date: Fri, 08 Apr 2011 19:59:23 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 257 v=0 o=root 1103328122 1103328122 IN IP4 172.30.1.47 s=Asterisk PBX 1.8.3.2 c=IN IP4 172.30.1.47 t=0 0 m=audio 12384 RTP/AVP 0 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Apr 8 12:59:27] WARNING[13920]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x2f32e00 (len 789) to 0.0.29.200:5060 returned -1: Invalid argument [Apr 8 12:59:27] NOTICE[13917]: chan_iax2.c:4643 __auto_congest: Auto-congesting call due to slow response -- IAX2/0.0.29.200:4569-4965 is circuit-busy -- Hungup 'IAX2/0.0.29.200:4569-4965' Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 000 Type: IAX Subclass: HANGUP Timestamp: 04005ms SCall: 04965 DCall: 00000 [0.0.29.200:4569] CAUSE CODE : 0 Tx-Frame Retry[001] -- OSeqno: 001 ISeqno: 000 Type: IAX Subclass: HANGUP Timestamp: 04005ms SCall: 04965 DCall: 00000 [0.0.29.200:4569] CAUSE CODE : 0 Retransmitting #4 (no NAT) to 0.0.29.200:5060: INVITE sip:7624 SIP/2.0 Via: SIP/2.0/UDP 172.30.1.47:5060;branch=z9hG4bK1789e45d Max-Forwards: 70 From: "Cambridge Guest" ;tag=as1f3e09eb To: Contact: Call-ID: 6d8be7981c3219884243c61b4a4de639@172.30.1.47:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.3.2 Date: Fri, 08 Apr 2011 19:59:23 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 257 v=0 o=root 1103328122 1103328122 IN IP4 172.30.1.47 s=Asterisk PBX 1.8.3.2 c=IN IP4 172.30.1.47 t=0 0 m=audio 12384 RTP/AVP 0 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Apr 8 12:59:31] WARNING[13920]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x2f32e00 (len 789) to 0.0.29.200:5060 returned -1: Invalid argument Tx-Frame Retry[002] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 00005ms SCall: 04965 DCall: 00000 [0.0.29.200:4569] VERSION : 2 CALLED NUMBER : s CODEC_PREFS : () CALLING NUMBER : 7527 CALLING PRESNTN : 0 CALLING TYPEOFN : 0 CALLING TRANSIT : 0 CALLING NAME : Cambridge Guest LANGUAGE : en FORMAT : 2 FORMAT2 : gsm CAPABILITY : 1795 CAPABILITY2 : unknown ADSICPE : 2 DATE TIME : 2011-04-08 12:59:22 Retransmitting #5 (no NAT) to 0.0.29.200:5060: INVITE sip:7624 SIP/2.0 Via: SIP/2.0/UDP 172.30.1.47:5060;branch=z9hG4bK1789e45d Max-Forwards: 70 From: "Cambridge Guest" ;tag=as1f3e09eb To: Contact: Call-ID: 6d8be7981c3219884243c61b4a4de639@172.30.1.47:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.3.2 Date: Fri, 08 Apr 2011 19:59:23 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 257 v=0 o=root 1103328122 1103328122 IN IP4 172.30.1.47 s=Asterisk PBX 1.8.3.2 c=IN IP4 172.30.1.47 t=0 0 m=audio 12384 RTP/AVP 0 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Apr 8 12:59:39] WARNING[13920]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x2f32e00 (len 789) to 0.0.29.200:5060 returned -1: Invalid argument Reliably Transmitting (no NAT) to 172.30.254.48:5060: OPTIONS sip:7527@172.30.254.48 SIP/2.0 Via: SIP/2.0/UDP 172.30.1.47:5060;branch=z9hG4bK7dc3ffd7 Max-Forwards: 70 From: "asterisk" ;tag=as57dfe083 To: Contact: Call-ID: 77506e220482eae05f3dbdbd0b9dcb7c@172.30.1.47:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.3.2 Date: Fri, 08 Apr 2011 19:59:43 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- <--- SIP read from UDP:172.30.254.48:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.30.1.47:5060;branch=z9hG4bK7dc3ffd7 From: "asterisk" ;tag=as57dfe083 To: ;tag=BE6D0D1A-BD7D95D5 CSeq: 102 OPTIONS Call-ID: 77506e220482eae05f3dbdbd0b9dcb7c@172.30.1.47:5060 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.5.0043 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Really destroying SIP dialog '77506e220482eae05f3dbdbd0b9dcb7c@172.30.1.47:5060' Method: OPTIONS -- Nobody picked up in 20000 ms Scheduling destruction of SIP dialog '6d8be7981c3219884243c61b4a4de639@172.30.1.47:5060' in 32000 ms (Method: INVITE) -- Executing [s@macro-stdexten:2] Goto("SIP/7527-000000d8", "s-NOANSWER,1") in new stack -- Goto (macro-stdexten,s-NOANSWER,1) -- Executing [s-NOANSWER@macro-stdexten:1] VoiceMail("SIP/7527-000000d8", "7624,u") in new stack Audio is at 5060 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 172.30.254.48:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.30.254.48;branch=z9hG4bK2f2634f4C8756E5D;received=172.30.254.48 From: "Cambridge Guest" ;tag=DA87FFEE-A33412B9 To: ;tag=as6819eff0 Call-ID: 47e59e82-4cf00900-7c1efcf3@172.30.254.48 CSeq: 1 INVITE Server: Asterisk PBX 1.8.3.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 234 v=0 o=root 1513891678 1513891678 IN IP4 172.30.1.47 s=Asterisk PBX 1.8.3.2 c=IN IP4 172.30.1.47 t=0 0 m=audio 19680 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 <------------> <--- SIP read from UDP:172.30.254.48:5060 ---> ACK sip:7624@172.30.1.47:5060 SIP/2.0 Via: SIP/2.0/UDP 172.30.254.48;branch=z9hG4bKd83fe22bD9972658 From: "Cambridge Guest" ;tag=DA87FFEE-A33412B9 To: ;tag=as6819eff0 CSeq: 1 ACK Call-ID: 47e59e82-4cf00900-7c1efcf3@172.30.254.48 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.5.0043 Max-Forwards: 70 Content-Length: 0 <-------------> --- (11 headers 0 lines) --- [Apr 8 12:59:44] WARNING[15249]: app_voicemail.c:5535 leave_voicemail: No entry in voicemail config file for '7624' -- Executing [s-NOANSWER@macro-stdexten:2] Goto("SIP/7527-000000d8", "default,s,1") in new stack -- Goto (default,s,1) == Channel 'SIP/7527-000000d8' jumping out of macro 'stdexten' -- Sent into invalid extension 's' in context 'default' on SIP/7527-000000d8 -- Executing [i@default:1] Playback("SIP/7527-000000d8", "invalid") in new stack -- Playing 'invalid.ulaw' (language 'en') Reliably Transmitting (no NAT) to 172.30.245.71:5060: OPTIONS sip:7516@172.30.245.71 SIP/2.0 Via: SIP/2.0/UDP 172.30.1.47:5060;branch=z9hG4bK2a3d392b Max-Forwards: 70 From: "asterisk" ;tag=as2a39f1f5 To: Contact: Call-ID: 1dea34bf10da195977eb50966571b306@172.30.1.47:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.3.2 Date: Fri, 08 Apr 2011 19:59:45 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- <--- SIP read from UDP:172.30.245.71:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.30.1.47:5060;branch=z9hG4bK2a3d392b From: "asterisk" ;tag=as2a39f1f5 To: ;tag=3AE1872E-EE021A8B CSeq: 102 OPTIONS Call-ID: 1dea34bf10da195977eb50966571b306@172.30.1.47:5060 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.5.0043 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Really destroying SIP dialog '1dea34bf10da195977eb50966571b306@172.30.1.47:5060' Method: OPTIONS -- Executing [i@default:2] Hangup("SIP/7527-000000d8", "") in new stack == Spawn extension (default, i, 2) exited non-zero on 'SIP/7527-000000d8' Scheduling destruction of SIP dialog '47e59e82-4cf00900-7c1efcf3@172.30.254.48' in 6400 ms (Method: ACK) set_destination: Parsing for address/port to send to set_destination: set destination to 172.30.254.48:5060 Reliably Transmitting (no NAT) to 172.30.254.48:5060: BYE sip:7527@172.30.254.48 SIP/2.0 Via: SIP/2.0/UDP 172.30.1.47:5060;branch=z9hG4bK1672f8f9 Max-Forwards: 70 From: ;tag=as6819eff0 To: "Cambridge Guest" ;tag=DA87FFEE-A33412B9 Call-ID: 47e59e82-4cf00900-7c1efcf3@172.30.254.48 CSeq: 102 BYE User-Agent: Asterisk PBX 1.8.3.2 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- <--- SIP read from UDP:172.30.254.48:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.30.1.47:5060;branch=z9hG4bK1672f8f9 From: ;tag=as6819eff0 To: "Cambridge Guest" ;tag=DA87FFEE-A33412B9 CSeq: 102 BYE Call-ID: 47e59e82-4cf00900-7c1efcf3@172.30.254.48 Contact: User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.5.0043 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- SIP Response message for INCOMING dialog BYE arrived Really destroying SIP dialog '47e59e82-4cf00900-7c1efcf3@172.30.254.48' Method: ACK Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ Timestamp: 00006ms SCall: 02093 DCall: 00000 [172.30.245.208:4569] USERNAME : orasebcam REFRESH : 60 Rx-Frame Retry[Yes] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: CTOKEN Timestamp: 00006ms SCall: 00001 DCall: 02093 [172.30.245.208:4569] CALLTOKEN : 51 bytes Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ Timestamp: 00009ms SCall: 02093 DCall: 00000 [172.30.245.208:4569] USERNAME : orasebcam REFRESH : 60 CALLTOKEN : 51 bytes Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REGAUTH Timestamp: 00019ms SCall: 15741 DCall: 02093 [172.30.245.208:4569] AUTHMETHODS : 2 CHALLENGE : \x31\x30\x35\x39\x34\x30\x37\x38\x37 USERNAME : orasebcam Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: REGREQ Timestamp: 00012ms SCall: 02093 DCall: 15741 [172.30.245.208:4569] USERNAME : orasebcam REFRESH : 60 MD5 RESULT : 63d3cf33db5fcc7a3bc2d6153fd16953 Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: REGACK Timestamp: 00020ms SCall: 15741 DCall: 02093 [172.30.245.208:4569] USERNAME : orasebcam DATE TIME : 2011-04-08 13:01:34 REFRESH : 60 APPARENT ADDRES : IPV4 172.30.1.47:4569 Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: ACK Timestamp: 00020ms SCall: 02093 DCall: 15741 [172.30.245.208:4569] Retransmitting #6 (no NAT) to 0.0.29.200:5060: INVITE sip:7624 SIP/2.0 Via: SIP/2.0/UDP 172.30.1.47:5060;branch=z9hG4bK1789e45d Max-Forwards: 70 From: "Cambridge Guest" ;tag=as1f3e09eb To: Contact: Call-ID: 6d8be7981c3219884243c61b4a4de639@172.30.1.47:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.3.2 Date: Fri, 08 Apr 2011 19:59:23 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 257 v=0 o=root 1103328122 1103328122 IN IP4 172.30.1.47 s=Asterisk PBX 1.8.3.2 c=IN IP4 172.30.1.47 t=0 0 m=audio 12384 RTP/AVP 0 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Apr 8 12:59:55] WARNING[13920]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x2f32e00 (len 789) to 0.0.29.200:5060 returned -1: Invalid argument [Apr 8 12:59:55] WARNING[13920]: chan_sip.c:3386 retrans_pkt: Retransmission timeout reached on transmission 6d8be7981c3219884243c61b4a4de639@172.30.1.47:5060 for seqno 102 (Critical Request) -- See doc/sip-retransmit.txt. Packet timed out after 32000ms with no response Really destroying SIP dialog '6d8be7981c3219884243c61b4a4de639@172.30.1.47:5060' Method: INVITE shirley*CLI> Disconnected from Asterisk server root@shirley:~# asterisk -r Verbosity is at least 15 Core debug is at least 15 shirley*CLI> core set verbose 0 Verbosity is now OFF shirley*CLI> core set debug 0 Core debug is now OFF shirley*CLI> sip set debug off SIP Debugging Disabled shirley*CLI> iax2 set debug off IAX2 Debugging Disabled shirley*CLI> exit