v=0 o=root 4259 4259 IN IP4 192.168.100.150 s=session c=IN IP4 192.168.100.150 t=0 0 m=audio 16614 RTP/AVP 98 101 a=rtpmap:98 iLBC/8000 a=fmtp:98 mode=30 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:30 a=sendrecv --- [May 7 18:09:49] DEBUG[4338]: channel.c:1908 ast_settimeout: Scheduling timer at 0 sample intervals [May 7 18:09:49] DEBUG[4338]: channel.c:1908 ast_settimeout: Scheduling timer at 0 sample intervals Retransmitting #4 (NAT) to PublicIPUser:22526: OPTIONS sip:60006@PublicIPUser:17282;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.100.150:5060;branch=z9hG4bK5eccf5ae;rport From: "asterisk" ;tag=as12982cee To: Contact: Call-ID: 32ff62b525b28608524ac95a20f87b79@192.168.100.150 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 07 May 2007 16:09:45 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- Really destroying SIP dialog '32ff62b525b28608524ac95a20f87b79@192.168.100.150' Method: OPTIONS xasterisk*CLI> <--- SIP read from PublicIPUser:14875 ---> <-------------> --- (0 headers 1 lines) --- Retransmitting #3 (NAT) to PublicIPUser:14875: SIP/2.0 200 OK Via: SIP/2.0/UDP PublicIPUser:14542;branch=z9hG4bK-d87543-e53848525a493d45-1--d87543-;received=PublicIPUser;rport=14875 From: "Versia2";tag=a0408b15 To: "85000";tag=as2035457b Call-ID: ODVmY2VmNWM3OWM2N2EzODQ5M2QzOGE3ZDIzYjNhOTA. CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 265 v=0 o=root 4259 4259 IN IP4 192.168.100.150 s=session c=IN IP4 192.168.100.150 t=0 0 m=audio 16614 RTP/AVP 98 101 a=rtpmap:98 iLBC/8000 a=fmtp:98 mode=30 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:30 a=sendrecv --- xasterisk*CLI> <--- SIP read from 10.2.32.107:16484 ---> <-------------> --- (0 headers 1 lines) --- xasterisk*CLI> <--- SIP read from 10.32.3.120:24760 ---> <-------------> --- (0 headers 1 lines) --- Retransmitting #4 (NAT) to PublicIPUser:14875: SIP/2.0 200 OK Via: SIP/2.0/UDP PublicIPUser:14542;branch=z9hG4bK-d87543-e53848525a493d45-1--d87543-;received=PublicIPUser;rport=14875 From: "Versia2";tag=a0408b15 To: "85000";tag=as2035457b Call-ID: ODVmY2VmNWM3OWM2N2EzODQ5M2QzOGE3ZDIzYjNhOTA. CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 265 v=0 o=root 4259 4259 IN IP4 192.168.100.150 s=session c=IN IP4 192.168.100.150 t=0 0 m=audio 16614 RTP/AVP 98 101 a=rtpmap:98 iLBC/8000 a=fmtp:98 mode=30 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:30 a=sendrecv --- xasterisk*CLI> <--- SIP read from 10.2.12.240:57644 ---> <-------------> --- (0 headers 1 lines) --- Retransmitting #5 (NAT) to PublicIPUser:14875: SIP/2.0 200 OK Via: SIP/2.0/UDP PublicIPUser:14542;branch=z9hG4bK-d87543-e53848525a493d45-1--d87543-;received=PublicIPUser;rport=14875 From: "Versia2";tag=a0408b15 To: "85000";tag=as2035457b Call-ID: ODVmY2VmNWM3OWM2N2EzODQ5M2QzOGE3ZDIzYjNhOTA. CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 265 v=0 o=root 4259 4259 IN IP4 192.168.100.150 s=session c=IN IP4 192.168.100.150 t=0 0 m=audio 16614 RTP/AVP 98 101 a=rtpmap:98 iLBC/8000 a=fmtp:98 mode=30 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:30 a=sendrecv --- [May 7 18:09:59] DEBUG[4338]: app_voicemail.c:6028 vm_authenticate: Before find user for mailbox 60004 [May 7 18:09:59] DEBUG[4338]: app_voicemail.c:6202 vm_execmain: After vm_authenticate [May 7 18:09:59] DEBUG[4338]: app_voicemail.c:6237 vm_execmain: Before open_mailbox [May 7 18:09:59] DEBUG[4338]: app.c:938 ast_lock_path: Locked path '/var/spool/asterisk/voicemail/default/60004/Old' [May 7 18:09:59] DEBUG[4338]: app.c:958 ast_unlock_path: Unlocked path '/var/spool/asterisk/voicemail/default/60004/Old' [May 7 18:09:59] DEBUG[4338]: app.c:938 ast_lock_path: Locked path '/var/spool/asterisk/voicemail/default/60004/Old' [May 7 18:09:59] DEBUG[4338]: app.c:958 ast_unlock_path: Unlocked path '/var/spool/asterisk/voicemail/default/60004/Old' [May 7 18:09:59] DEBUG[4338]: app_voicemail.c:6242 vm_execmain: Number of old messages: 1 [May 7 18:09:59] DEBUG[4338]: app.c:938 ast_lock_path: Locked path '/var/spool/asterisk/voicemail/default/60004/INBOX' [May 7 18:09:59] DEBUG[4338]: app.c:958 ast_unlock_path: Unlocked path '/var/spool/asterisk/voicemail/default/60004/INBOX' [May 7 18:09:59] DEBUG[4338]: app.c:938 ast_lock_path: Locked path '/var/spool/asterisk/voicemail/default/60004/INBOX' [May 7 18:09:59] DEBUG[4338]: app.c:958 ast_unlock_path: Unlocked path '/var/spool/asterisk/voicemail/default/60004/INBOX' [May 7 18:09:59] DEBUG[4338]: app_voicemail.c:6248 vm_execmain: Number of new messages: 8 [May 7 18:09:59] DEBUG[4338]: channel.c:1908 ast_settimeout: Scheduling timer at 160 sample intervals -- Playing 'vm-youhave' (language 'en') [May 7 18:09:59] DEBUG[4290]: acl.c:213 ast_apply_ha: ##### Testing PublicIPUser with 10.1.0.0 [May 7 18:09:59] DEBUG[4290]: acl.c:213 ast_apply_ha: ##### Testing PublicIPUser with 10.2.0.0 [May 7 18:09:59] DEBUG[4290]: acl.c:213 ast_apply_ha: ##### Testing 192.168.100.150 with 10.1.0.0 [May 7 18:09:59] DEBUG[4290]: acl.c:213 ast_apply_ha: ##### Testing 192.168.100.150 with 10.2.0.0 Reliably Transmitting (NAT) to PublicIPUser:22526: OPTIONS sip:60006@PublicIPUser:17282;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.100.150:5060;branch=z9hG4bK468a4afa;rport From: "asterisk" ;tag=as21075f86 To: Contact: Call-ID: 67fdbad73564fb7f4c3a87625097bece@192.168.100.150 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 07 May 2007 16:09:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [May 7 18:10:00] DEBUG[4338]: channel.c:1908 ast_settimeout: Scheduling timer at 0 sample intervals [May 7 18:10:00] DEBUG[4338]: channel.c:1908 ast_settimeout: Scheduling timer at 0 sample intervals [May 7 18:10:00] DEBUG[4338]: channel.c:1908 ast_settimeout: Scheduling timer at 160 sample intervals -- Playing 'digits/8' (language 'en') Retransmitting #1 (NAT) to PublicIPUser:22526: OPTIONS sip:60006@PublicIPUser:17282;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.100.150:5060;branch=z9hG4bK468a4afa;rport From: "asterisk" ;tag=as21075f86 To: Contact: Call-ID: 67fdbad73564fb7f4c3a87625097bece@192.168.100.150 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 07 May 2007 16:09:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [May 7 18:10:01] DEBUG[4338]: channel.c:1908 ast_settimeout: Scheduling timer at 0 sample intervals [May 7 18:10:01] DEBUG[4338]: channel.c:1908 ast_settimeout: Scheduling timer at 0 sample intervals [May 7 18:10:01] DEBUG[4338]: channel.c:1908 ast_settimeout: Scheduling timer at 160 sample intervals -- Playing 'vm-INBOX' (language 'en') Retransmitting #2 (NAT) to PublicIPUser:22526: OPTIONS sip:60006@PublicIPUser:17282;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.100.150:5060;branch=z9hG4bK468a4afa;rport From: "asterisk" ;tag=as21075f86 To: Contact: Call-ID: 67fdbad73564fb7f4c3a87625097bece@192.168.100.150 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 07 May 2007 16:09:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [May 7 18:10:01] DEBUG[4338]: channel.c:1908 ast_settimeout: Scheduling timer at 0 sample intervals [May 7 18:10:01] DEBUG[4338]: channel.c:1908 ast_settimeout: Scheduling timer at 0 sample intervals [May 7 18:10:01] DEBUG[4338]: channel.c:1908 ast_settimeout: Scheduling timer at 160 sample intervals -- Playing 'vm-and' (language 'en') [May 7 18:10:02] DEBUG[4338]: channel.c:1908 ast_settimeout: Scheduling timer at 0 sample intervals [May 7 18:10:02] DEBUG[4338]: channel.c:1908 ast_settimeout: Scheduling timer at 0 sample intervals [May 7 18:10:02] DEBUG[4338]: channel.c:1908 ast_settimeout: Scheduling timer at 160 sample intervals -- Playing 'digits/1' (language 'en') Retransmitting #3 (NAT) to PublicIPUser:22526: OPTIONS sip:60006@PublicIPUser:17282;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.100.150:5060;branch=z9hG4bK468a4afa;rport From: "asterisk" ;tag=as21075f86 To: Contact: Call-ID: 67fdbad73564fb7f4c3a87625097bece@192.168.100.150 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 07 May 2007 16:09:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- Retransmitting #6 (NAT) to PublicIPUser:14875: SIP/2.0 200 OK Via: SIP/2.0/UDP PublicIPUser:14542;branch=z9hG4bK-d87543-e53848525a493d45-1--d87543-;received=PublicIPUser;rport=14875 From: "Versia2";tag=a0408b15 To: "85000";tag=as2035457b Call-ID: ODVmY2VmNWM3OWM2N2EzODQ5M2QzOGE3ZDIzYjNhOTA. CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 265 v=0 o=root 4259 4259 IN IP4 192.168.100.150 s=session c=IN IP4 192.168.100.150 t=0 0 m=audio 16614 RTP/AVP 98 101 a=rtpmap:98 iLBC/8000 a=fmtp:98 mode=30 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:30 a=sendrecv --- [May 7 18:10:03] DEBUG[4338]: channel.c:1908 ast_settimeout: Scheduling timer at 0 sample intervals [May 7 18:10:03] DEBUG[4338]: channel.c:1908 ast_settimeout: Scheduling timer at 0 sample intervals [May 7 18:10:03] DEBUG[4338]: channel.c:1908 ast_settimeout: Scheduling timer at 160 sample intervals -- Playing 'vm-Old' (language 'en') Retransmitting #4 (NAT) to PublicIPUser:22526: OPTIONS sip:60006@PublicIPUser:17282;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.100.150:5060;branch=z9hG4bK468a4afa;rport From: "asterisk" ;tag=as21075f86 To: Contact: Call-ID: 67fdbad73564fb7f4c3a87625097bece@192.168.100.150 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 07 May 2007 16:09:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- Really destroying SIP dialog '67fdbad73564fb7f4c3a87625097bece@192.168.100.150' Method: OPTIONS [May 7 18:10:04] DEBUG[4338]: channel.c:1908 ast_settimeout: Scheduling timer at 0 sample intervals [May 7 18:10:04] DEBUG[4338]: channel.c:1908 ast_settimeout: Scheduling timer at 0 sample intervals [May 7 18:10:04] DEBUG[4338]: channel.c:1908 ast_settimeout: Scheduling timer at 160 sample intervals -- Playing 'vm-message' (language 'en') [May 7 18:10:05] DEBUG[4338]: channel.c:1908 ast_settimeout: Scheduling timer at 0 sample intervals [May 7 18:10:05] DEBUG[4338]: channel.c:1908 ast_settimeout: Scheduling timer at 0 sample intervals [May 7 18:10:05] DEBUG[4338]: channel.c:1908 ast_settimeout: Scheduling timer at 160 sample intervals -- Playing 'vm-onefor' (language 'en') [May 7 18:10:06] DEBUG[4338]: channel.c:1908 ast_settimeout: Scheduling timer at 0 sample intervals [May 7 18:10:06] DEBUG[4338]: channel.c:1908 ast_settimeout: Scheduling timer at 0 sample intervals [May 7 18:10:06] DEBUG[4338]: channel.c:1908 ast_settimeout: Scheduling timer at 160 sample intervals -- Playing 'vm-INBOX' (language 'en') [May 7 18:10:06] WARNING[4290]: chan_sip.c:1881 retrans_pkt: Maximum retries exceeded on transmission ODVmY2VmNWM3OWM2N2EzODQ5M2QzOGE3ZDIzYjNhOTA. for seqno 1 (Critical Response) [May 7 18:10:06] WARNING[4290]: chan_sip.c:1898 retrans_pkt: Hanging up call ODVmY2VmNWM3OWM2N2EzODQ5M2QzOGE3ZDIzYjNhOTA. - no reply to our critical packet. [May 7 18:10:06] DEBUG[4338]: channel.c:1908 ast_settimeout: Scheduling timer at 0 sample intervals [May 7 18:10:06] DEBUG[4338]: app.c:938 ast_lock_path: Locked path '/var/spool/asterisk/voicemail/default/60004/INBOX' [May 7 18:10:06] DEBUG[4338]: app.c:958 ast_unlock_path: Unlocked path '/var/spool/asterisk/voicemail/default/60004/INBOX' == Spawn extension (default, 85000, 1) exited non-zero on 'SIP/PublicIP*-08216808' [May 7 18:10:06] DEBUG[4338]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is '"Versia2" <60004>' [May 7 18:10:06] DEBUG[4338]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is '60004' [May 7 18:10:06] DEBUG[4338]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is '85000' [May 7 18:10:06] DEBUG[4338]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is 'default' [May 7 18:10:06] DEBUG[4338]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is 'SIP/PublicIP*-08216808' [May 7 18:10:06] DEBUG[4338]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is '' [May 7 18:10:06] DEBUG[4338]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is 'VoiceMailMain' [May 7 18:10:06] DEBUG[4338]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is '' [May 7 18:10:06] DEBUG[4338]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is '2007-05-07 18:09:46' [May 7 18:10:06] DEBUG[4338]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is '2007-05-07 18:09:46' [May 7 18:10:06] DEBUG[4338]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is '2007-05-07 18:10:06' [May 7 18:10:06] DEBUG[4338]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is '20' [May 7 18:10:06] DEBUG[4338]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is '20' [May 7 18:10:06] DEBUG[4338]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is 'ANSWERED' [May 7 18:10:06] DEBUG[4338]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is 'DOCUMENTATION' [May 7 18:10:06] DEBUG[4338]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is '' [May 7 18:10:06] DEBUG[4338]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is '1178554186.2' [May 7 18:10:06] DEBUG[4338]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is '' Really destroying SIP dialog 'ODVmY2VmNWM3OWM2N2EzODQ5M2QzOGE3ZDIzYjNhOTA.' Method: INVITE [May 7 18:10:13] DEBUG[4290]: acl.c:213 ast_apply_ha: ##### Testing PublicIPUser with 10.1.0.0 [May 7 18:10:13] DEBUG[4290]: acl.c:213 ast_apply_ha: ##### Testing PublicIPUser with 10.2.0.0 [May 7 18:10:13] DEBUG[4290]: acl.c:213 ast_apply_ha: ##### Testing 192.168.100.150 with 10.1.0.0 [May 7 18:10:13] DEBUG[4290]: acl.c:213 ast_apply_ha: ##### Testing 192.168.100.150 with 10.2.0.0 Reliably Transmitting (NAT) to PublicIPUser:22526: OPTIONS sip:60006@PublicIPUser:17282;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.100.150:5060;branch=z9hG4bK1aa0355f;rport From: "asterisk" ;tag=as70a0f7fa To: Contact: Call-ID: 4af6c2401b082a2559508aee5ce5f132@192.168.100.150 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 07 May 2007 16:10:13 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- Retransmitting #1 (NAT) to PublicIPUser:22526: OPTIONS sip:60006@PublicIPUser:17282;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.100.150:5060;branch=z9hG4bK1aa0355f;rport From: "asterisk" ;tag=as70a0f7fa To: Contact: Call-ID: 4af6c2401b082a2559508aee5ce5f132@192.168.100.150 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 07 May 2007 16:10:13 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- Retransmitting #2 (NAT) to PublicIPUser:22526: OPTIONS sip:60006@PublicIPUser:17282;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.100.150:5060;branch=z9hG4bK1aa0355f;rport From: "asterisk" ;tag=as70a0f7fa To: Contact: Call-ID: 4af6c2401b082a2559508aee5ce5f132@192.168.100.150 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 07 May 2007 16:10:13 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- Retransmitting #3 (NAT) to PublicIPUser:22526: OPTIONS sip:60006@PublicIPUser:17282;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.100.150:5060;branch=z9hG4bK1aa0355f;rport From: "asterisk" ;tag=as70a0f7fa To: Contact: Call-ID: 4af6c2401b082a2559508aee5ce5f132@192.168.100.150 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 07 May 2007 16:10:13 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- Retransmitting #4 (NAT) to PublicIPUser:22526: OPTIONS sip:60006@PublicIPUser:17282;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.100.150:5060;branch=z9hG4bK1aa0355f;rport From: "asterisk" ;tag=as70a0f7fa To: Contact: Call-ID: 4af6c2401b082a2559508aee5ce5f132@192.168.100.150 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 07 May 2007 16:10:13 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- Really destroying SIP dialog '4af6c2401b082a2559508aee5ce5f132@192.168.100.150' Method: OPTIONS xasterisk*CLI> <--- SIP read from 10.40.8.101:6230 ---> <-------------> --- (0 headers 1 lines) --- xasterisk*CLI> <--- SIP read from PublicIPUser:14875 ---> <-------------> --- (0 headers 1 lines) --- xasterisk*CLI> <--- SIP read from 10.2.32.107:16484 ---> <-------------> --- (0 headers 1 lines) --- xasterisk*CLI> <--- SIP read from 10.32.3.120:24760 ---> <-------------> --- (0 headers 1 lines) --- xasterisk*CLI> xasterisk*CLI> <--- SIP read from 10.2.12.240:57644 ---> <-------------> --- (0 headers 1 lines) --- [May 7 18:10:27] DEBUG[4290]: acl.c:213 ast_apply_ha: ##### Testing PublicIPUser with 10.1.0.0 [May 7 18:10:27] DEBUG[4290]: acl.c:213 ast_apply_ha: ##### Testing PublicIPUser with 10.2.0.0 [May 7 18:10:27] DEBUG[4290]: acl.c:213 ast_apply_ha: ##### Testing 192.168.100.150 with 10.1.0.0 [May 7 18:10:27] DEBUG[4290]: acl.c:213 ast_apply_ha: ##### Testing 192.168.100.150 with 10.2.0.0 Reliably Transmitting (NAT) to PublicIPUser:22526: OPTIONS sip:60006@PublicIPUser:17282;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.100.150:5060;branch=z9hG4bK015a8d53;rport From: "asterisk" ;tag=as13c7b14f To: Contact: Call-ID: 39b81caa76fe09886df738dd5c1c9f90@192.168.100.150 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 07 May 2007 16:10:27 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- xasterisk*CLI> sip no debug SIP Debugging Disabled [May 7 18:10:41] DEBUG[4290]: acl.c:213 ast_apply_ha: ##### Testing PublicIPUser with 10.1.0.0 [May 7 18:10:41] DEBUG[4290]: acl.c:213 ast_apply_ha: ##### Testing PublicIPUser with 10.2.0.0 [May 7 18:10:41] DEBUG[4290]: acl.c:213 ast_apply_ha: ##### Testing 192.168.100.150 with 10.1.0.0 [May 7 18:10:41] DEBUG[4290]: acl.c:213 ast_apply_ha: ##### Testing 192.168.100.150 with 10.2.0.0 [May 7 18:10:45] DEBUG[4290]: acl.c:213 ast_apply_ha: ##### Testing PublicIPUser with 10.1.0.0 [May 7 18:10:45] DEBUG[4290]: acl.c:213 ast_apply_ha: ##### Testing PublicIPUser with 10.2.0.0 [May 7 18:10:45] DEBUG[4290]: acl.c:213 ast_apply_ha: ##### Testing 192.168.100.150 with 10.1.0.0 [May 7 18:10:45] DEBUG[4290]: acl.c:213 ast_apply_ha: ##### Testing 192.168.100.150 with 10.2.0.0 xasterisk*CLI>