Current directory is /home/areyouwireless/asterisk/asterisk-1.6.2.1/main/ GNU gdb 6.4.90-debian Copyright (C) 2006 Free Software Foundation, Inc. GDB is free software, covered by the GNU General Public License, and you are welcome to change it and/or distribute copies of it under certain conditions. Type "show copying" to see the conditions. There is absolutely no warranty for GDB. Type "show warranty" for details. This GDB was configured as "i486-linux-gnu"...Using host libthread_db library "/lib/tls/i686/cmov/libthread_db.so.1". (gdb) run -c Starting program: /home/areyouwireless/asterisk/asterisk-1.6.2.1/main/asterisk -c [Thread debugging using libthread_db enabled] [New Thread -1213818432 (LWP 22227)] Asterisk 1.6.2.1, Copyright (C) 1999 - 2009 Digium, Inc. and others. Created by Mark Spencer Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. ========================================================================= [ Booting... [ Reading Master Configuration ] [ Initializing Custom Configuration Options ] [New Thread -1213822048 (LWP 22232)] [New Thread -1214067808 (LWP 22233)] [New Thread -1214313568 (LWP 22234)] [New Thread -1214694496 (LWP 22235)] [Jan 30 09:39:51] NOTICE[22227]: cdr.c:1473 do_reload: CDR simple logging enabled. [New Thread -1214940256 (LWP 22236)] [New Thread -1215186016 (LWP 22237)] [New Thread -1215431776 (LWP 22238)] [Jan 30 09:39:51] NOTICE[22227]: loader.c:1044 load_modules: 162 modules will be loaded. ..[Jan 30 09:39:53] NOTICE[22227]: res_smdi.c:1361 load_module: No SMDI interfaces are available to listen on, not starting SMDI listener. .....[New Thread -1221309536 (LWP 22239)] .......................[New Thread -1221555296 (LWP 22240)] .................................[New Thread -1221887072 (LWP 22241)] ..............[Jan 30 09:40:00] WARNING[22227]: translate.c:654 __ast_register_translator: plc_samples 160 format f .........[New Thread -1222132832 (LWP 22242)] ....................[New Thread -1222378592 (LWP 22243)] ........[Jan 30 09:40:00] NOTICE[22227]: pbx_ael.c:122 pbx_load_module: Starting AEL load process. [Jan 30 09:40:00] NOTICE[22227]: pbx_ael.c:135 pbx_load_module: AEL load process: parsed config file name '/etc/asterisk/extensions.ael'. [Jan 30 09:40:00] NOTICE[22227]: pbx_ael.c:138 pbx_load_module: AEL load process: checked config file name '/etc/asterisk/extensions.ael'. [Jan 30 09:40:00] NOTICE[22227]: pbx_ael.c:141 pbx_load_module: AEL load process: compiled config file name '/etc/asterisk/extensions.ael'. [Jan 30 09:40:00] NOTICE[22227]: pbx_ael.c:146 pbx_load_module: AEL load process: merged config file name '/etc/asterisk/extensions.ael'. [Jan 30 09:40:00] NOTICE[22227]: pbx_ael.c:149 pbx_load_module: AEL load process: verified config file name '/etc/asterisk/extensions.ael'. ...[New Thread -1222624352 (LWP 22244)] ..................... == Aliased CLI command 'hangup request' to 'channel request hangup' == Aliased CLI command 'originate' to 'channel originate' == Aliased CLI command 'help' to 'core show help' == Aliased CLI command 'pri intense debug span' to 'pri set debug 2 span' == Aliased CLI command 'reload' to 'module reload' .....SIP channel loading... [Jan 30 09:40:00] NOTICE[22227]: chan_sip.c:23853 build_peer: The 'username' field for sip peers has been deprecated in favor of the term 'defaultuser' [New Thread -1222870112 (LWP 22245)] ..[New Thread -1223115872 (LWP 22246)] ...[New Thread -1223361632 (LWP 22247)] [New Thread -1223607392 (LWP 22248)] [New Thread -1223853152 (LWP 22249)] ....[Jan 30 09:40:01] NOTICE[22227]: chan_skinny.c:7062 config_load: Configuring skinny from skinny.conf [New Thread -1224098912 (LWP 22250)] [New Thread -1224344672 (LWP 22251)] .[New Thread -1225122912 (LWP 22252)] [Jan 30 09:40:01] WARNING[22227]: utils.c:1536 __ast_string_field_init: trying to reset empty pool [Jan 30 09:40:01] WARNING[22227]: utils.c:1536 __ast_string_field_init: trying to reset empty pool [Jan 30 09:40:01] WARNING[22227]: utils.c:1536 __ast_string_field_init: trying to reset empty pool [New Thread -1225368672 (LWP 22253)] [New Thread -1225614432 (LWP 22254)] [New Thread -1225860192 (LWP 22255)] [New Thread -1226105952 (LWP 22256)] [New Thread -1226351712 (LWP 22257)] [New Thread -1226597472 (LWP 22258)] [New Thread -1226843232 (LWP 22259)] [New Thread -1227088992 (LWP 22260)] [New Thread -1227334752 (LWP 22261)] [New Thread -1227580512 (LWP 22262)] [New Thread -1227826272 (LWP 22263)] .[New Thread -1228072032 (LWP 22264)] ....... ] Asterisk Ready. [New Thread -1228317792 (LWP 22265)] *CLI> memory show summary memory show summary 484 bytes in 1 allocations in file 'app_confbridge.c' 304 bytes in 2 allocations in file 'hashtab.c' 752 bytes in 2 allocations in file 'res_crypto.c' 1 bytes in 1 allocations in file 'app_dial.c' 936 bytes in 1 allocations in file 'ssl.c' 76 bytes in 1 allocations in file 'pbx_ael.c' 484 bytes in 1 allocations in file 'bridge_multiplexed.c' 244 bytes in 1 allocations in file 'iax2-provision.c' 14 bytes in 2 allocations in file 'http.c' 12324 bytes in 1 allocations in file 'stdtime/localtime.c' 548 bytes in 2 allocations in file 'features.c' 356 bytes in 1 allocations in file 'func_dialgroup.c' 312 bytes in 3 allocations in file '/home/areyouwireless/asterisk/asterisk-1.6.2.1/include/asterisk/threadsto' 484 bytes in 1 allocations in file 'app_queue.c' 4096 bytes in 1 allocations in file 'chan_unistim.c' 1704 bytes in 2 allocations in file 'chan_oss.c' 216 bytes in 2 allocations in file 'cdr.c' 9312 bytes in 3 allocations in file 'mpool/mpool.c' 4644 bytes in 2 allocations in file 'res_timing_pthread.c' 576 bytes in 4 allocations in file 'logger.c' 25440 bytes in 2 allocations in file 'app_followme.c' 638 bytes in 1 allocations in file '/home/areyouwireless/asterisk/asterisk-1.6.2.1/include/asterisk/strings.h' 212 bytes in 3 allocations in file 'netsock.c' 35963 bytes in 10 allocations in file 'app_minivm.c' 14055 bytes in 11 allocations in file 'event.c' 80 bytes in 10 allocations in file 'channel.c' 1395 bytes in 6 allocations in file 'res_clialiases.c' 20408 bytes in 12 allocations in file 'chan_sip.c' 84 bytes in 4 allocations in file 'timing.c' 969 bytes in 60 allocations in file 'ael/pval.c' 6980 bytes in 8 allocations in file 'app_voicemail.c' 30888 bytes in 24 allocations in file 'io.c' 779 bytes in 13 allocations in file 'taskprocessor.c' 4536 bytes in 21 allocations in file 'file.c' 1514 bytes in 8 allocations in file 'res_musiconhold.c' 23135 bytes in 297 allocations in file 'config.c' 10580 bytes in 45 allocations in file 'sched.c' 1832 bytes in 77 allocations in file 'manager.c' 96 bytes in 2 allocations in file 'devicestate.c' 44390 bytes in 527 allocations in file 'xmldoc.c' 33793 bytes in 361 allocations in file 'res_phoneprov.c' 2511 bytes in 34 allocations in file 'utils.c' 9584 bytes in 491 allocations in file 'cli.c' 39199 bytes in 1191 allocations in file 'indications.c' 14714 bytes in 486 allocations in file 'loader.c' 7821 bytes in 278 allocations in file 'asterisk.c' 341422 bytes in 1137 allocations in file 'pbx.c' 394704 bytes in 32892 allocations in file 'astobj2.c' 2077868 bytes in 32797 allocations in file 'chan_iax2.c' 3183457 bytes allocated in 70842 allocations *CLI> sip set debug on sip set debug on SIP Debugging enabled *CLI> <--- SIP read from UDP:10.211.55.2:5060 ---> REGISTER sip:10.211.55.3 SIP/2.0 Via: SIP/2.0/UDP 10.127.1.99:5060;rport;branch=z9hG4bKPj9D1D2FEA-0A8D-4C70-8291-9D65FAFA856D Max-Forwards: 70 From: ;tag=12DFAD59-75E5-49B7-9BC4-CAB12D6F12B3 To: Call-ID: 9845EE6A-E8AE-4EF2-922D-372B845142E2 CSeq: 48222 REGISTER User-Agent: PJSUA v1.5/i386-apple-darwin8.11.1 Contact: Expires: 300 Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Sending to 10.127.1.99 : 5060 (no NAT) <--- Transmitting (no NAT) to 10.127.1.99:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.127.1.99:5060;branch=z9hG4bKPj9D1D2FEA-0A8D-4C70-8291-9D65FAFA856D;received=10.211.55.2;rport=5060 From: ;tag=12DFAD59-75E5-49B7-9BC4-CAB12D6F12B3 To: ;tag=as2dbfae08 Call-ID: 9845EE6A-E8AE-4EF2-922D-372B845142E2 CSeq: 48222 REGISTER Server: Asterisk PBX 1.6.2.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="74c457f6" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '9845EE6A-E8AE-4EF2-922D-372B845142E2' in 32000 ms (Method: REGISTER) <--- SIP read from UDP:10.211.55.2:5060 ---> REGISTER sip:10.211.55.3 SIP/2.0 Via: SIP/2.0/UDP 10.127.1.99:5060;rport;branch=z9hG4bKPj3A5F53A7-4C0A-4FB1-B004-BB5E922DB609 Max-Forwards: 70 From: ;tag=12DFAD59-75E5-49B7-9BC4-CAB12D6F12B3 To: Call-ID: 9845EE6A-E8AE-4EF2-922D-372B845142E2 CSeq: 48223 REGISTER User-Agent: PJSUA v1.5/i386-apple-darwin8.11.1 Contact: Expires: 300 Authorization: Digest username="1001", realm="asterisk", nonce="74c457f6", uri="sip:10.211.55.3", response="ffa17938f96998f96c2ab7e86e732c5d", algorithm=MD5 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Sending to 10.127.1.99 : 5060 (no NAT) <--- Transmitting (no NAT) to 10.127.1.99:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.127.1.99:5060;branch=z9hG4bKPj3A5F53A7-4C0A-4FB1-B004-BB5E922DB609;received=10.211.55.2;rport=5060 From: ;tag=12DFAD59-75E5-49B7-9BC4-CAB12D6F12B3 To: ;tag=as2dbfae08 Call-ID: 9845EE6A-E8AE-4EF2-922D-372B845142E2 CSeq: 48223 REGISTER Server: Asterisk PBX 1.6.2.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Expires: 300 Contact: ;expires=300 Date: Sat, 30 Jan 2010 14:40:39 GMT Content-Length: 0 <------------> Scheduling destruction of SIP dialog '9845EE6A-E8AE-4EF2-922D-372B845142E2' in 32000 ms (Method: REGISTER) <--- SIP read from UDP:10.211.55.2:5060 ---> REGISTER sip:10.211.55.3 SIP/2.0 Via: SIP/2.0/UDP 10.127.1.99:5060;rport;branch=z9hG4bKPj88DE7A3D-C0E7-4DDC-95EB-8FEEDB7A334A Max-Forwards: 70 From: ;tag=B2840543-FF98-4D9D-8559-EC8A646D2C8E To: Call-ID: 9845EE6A-E8AE-4EF2-922D-372B845142E2 CSeq: 48224 REGISTER User-Agent: PJSUA v1.5/i386-apple-darwin8.11.1 Contact: Expires: 0 Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Sending to 10.127.1.99 : 5060 (no NAT) <--- Transmitting (no NAT) to 10.127.1.99:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.127.1.99:5060;branch=z9hG4bKPj88DE7A3D-C0E7-4DDC-95EB-8FEEDB7A334A;received=10.211.55.2;rport=5060 From: ;tag=B2840543-FF98-4D9D-8559-EC8A646D2C8E To: ;tag=as2dbfae08 Call-ID: 9845EE6A-E8AE-4EF2-922D-372B845142E2 CSeq: 48224 REGISTER Server: Asterisk PBX 1.6.2.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="179bbd65" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '9845EE6A-E8AE-4EF2-922D-372B845142E2' in 32000 ms (Method: REGISTER) <--- SIP read from UDP:10.211.55.2:5060 ---> REGISTER sip:10.211.55.3 SIP/2.0 Via: SIP/2.0/UDP 10.211.55.2:5060;rport;branch=z9hG4bKPj343A3A98-AD0D-4FC6-9AF6-5AD2C5F374B3 Max-Forwards: 70 From: ;tag=2B041451-4F4E-42E4-A753-12AECB81C55A To: Call-ID: 5ACEDF97-17FF-4028-A4C5-EDBE79D7BEED CSeq: 15061 REGISTER User-Agent: PJSUA v1.5/i386-apple-darwin8.11.1 Contact: Expires: 300 Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Sending to 10.211.55.2 : 5060 (no NAT) <--- Transmitting (no NAT) to 10.211.55.2:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.211.55.2:5060;branch=z9hG4bKPj343A3A98-AD0D-4FC6-9AF6-5AD2C5F374B3;received=10.211.55.2;rport=5060 From: ;tag=2B041451-4F4E-42E4-A753-12AECB81C55A To: ;tag=as023e179c Call-ID: 5ACEDF97-17FF-4028-A4C5-EDBE79D7BEED CSeq: 15061 REGISTER Server: Asterisk PBX 1.6.2.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3e8313aa" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '5ACEDF97-17FF-4028-A4C5-EDBE79D7BEED' in 32000 ms (Method: REGISTER) <--- SIP read from UDP:10.211.55.2:5060 ---> REGISTER sip:10.211.55.3 SIP/2.0 Via: SIP/2.0/UDP 10.211.55.2:5060;rport;branch=z9hG4bKPj61731462-31E9-48ED-8E81-C6D1E82C642C Max-Forwards: 70 From: ;tag=2B041451-4F4E-42E4-A753-12AECB81C55A To: Call-ID: 5ACEDF97-17FF-4028-A4C5-EDBE79D7BEED CSeq: 15062 REGISTER User-Agent: PJSUA v1.5/i386-apple-darwin8.11.1 Contact: Expires: 300 Authorization: Digest username="1001", realm="asterisk", nonce="3e8313aa", uri="sip:10.211.55.3", response="fa0d09429b4022b5e0ddb992f2fea2aa", algorithm=MD5 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Sending to 10.211.55.2 : 5060 (no NAT) <--- Transmitting (no NAT) to 10.211.55.2:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.211.55.2:5060;branch=z9hG4bKPj61731462-31E9-48ED-8E81-C6D1E82C642C;received=10.211.55.2;rport=5060 From: ;tag=2B041451-4F4E-42E4-A753-12AECB81C55A To: ;tag=as023e179c Call-ID: 5ACEDF97-17FF-4028-A4C5-EDBE79D7BEED CSeq: 15062 REGISTER Server: Asterisk PBX 1.6.2.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Expires: 300 Contact: ;expires=300 Date: Sat, 30 Jan 2010 14:40:39 GMT Content-Length: 0 <------------> Scheduling destruction of SIP dialog '5ACEDF97-17FF-4028-A4C5-EDBE79D7BEED' in 32000 ms (Method: REGISTER) <--- SIP read from UDP:10.211.55.2:5060 ---> REGISTER sip:10.211.55.3 SIP/2.0 Via: SIP/2.0/UDP 10.211.55.2:5060;rport;branch=z9hG4bKPj2BDFBE8B-0E78-4F6C-B963-58F32AB7194A Max-Forwards: 70 From: ;tag=B2840543-FF98-4D9D-8559-EC8A646D2C8E To: Call-ID: 9845EE6A-E8AE-4EF2-922D-372B845142E2 CSeq: 48225 REGISTER User-Agent: PJSUA v1.5/i386-apple-darwin8.11.1 Contact: Expires: 0 Authorization: Digest username="1001", realm="asterisk", nonce="179bbd65", uri="sip:10.211.55.3", response="a8abab11eeb3e637c4e551c85718df46", algorithm=MD5 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Sending to 10.211.55.2 : 5060 (no NAT) <--- Transmitting (no NAT) to 10.211.55.2:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.211.55.2:5060;branch=z9hG4bKPj2BDFBE8B-0E78-4F6C-B963-58F32AB7194A;received=10.211.55.2;rport=5060 From: ;tag=B2840543-FF98-4D9D-8559-EC8A646D2C8E To: ;tag=as2dbfae08 Call-ID: 9845EE6A-E8AE-4EF2-922D-372B845142E2 CSeq: 48225 REGISTER Server: Asterisk PBX 1.6.2.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Expires: 0 Date: Sat, 30 Jan 2010 14:40:39 GMT Content-Length: 0 <------------> Scheduling destruction of SIP dialog '9845EE6A-E8AE-4EF2-922D-372B845142E2' in 32000 ms (Method: REGISTER) *CLI> memory show summary memory show summary 484 bytes in 1 allocations in file 'app_confbridge.c' 304 bytes in 2 allocations in file 'hashtab.c' 752 bytes in 2 allocations in file 'res_crypto.c' 1 bytes in 1 allocations in file 'app_dial.c' 936 bytes in 1 allocations in file 'ssl.c' 76 bytes in 1 allocations in file 'pbx_ael.c' 484 bytes in 1 allocations in file 'bridge_multiplexed.c' 244 bytes in 1 allocations in file 'iax2-provision.c' 14 bytes in 2 allocations in file 'http.c' 12324 bytes in 1 allocations in file 'stdtime/localtime.c' 548 bytes in 2 allocations in file 'features.c' 356 bytes in 1 allocations in file 'func_dialgroup.c' 484 bytes in 1 allocations in file 'app_queue.c' 1132 bytes in 7 allocations in file '/home/areyouwireless/asterisk/asterisk-1.6.2.1/include/asterisk/threadsto' 4096 bytes in 1 allocations in file 'chan_unistim.c' 1704 bytes in 2 allocations in file 'chan_oss.c' 216 bytes in 2 allocations in file 'cdr.c' 9312 bytes in 3 allocations in file 'mpool/mpool.c' 4644 bytes in 2 allocations in file 'res_timing_pthread.c' 576 bytes in 4 allocations in file 'logger.c' 25440 bytes in 2 allocations in file 'app_followme.c' 1489 bytes in 2 allocations in file '/home/areyouwireless/asterisk/asterisk-1.6.2.1/include/asterisk/strings.h' 212 bytes in 3 allocations in file 'netsock.c' 35963 bytes in 10 allocations in file 'app_minivm.c' 14138 bytes in 13 allocations in file 'event.c' 80 bytes in 10 allocations in file 'channel.c' 1395 bytes in 6 allocations in file 'res_clialiases.c' 27276 bytes in 18 allocations in file 'chan_sip.c' 84 bytes in 4 allocations in file 'timing.c' 969 bytes in 60 allocations in file 'ael/pval.c' 6980 bytes in 8 allocations in file 'app_voicemail.c' 30888 bytes in 24 allocations in file 'io.c' 779 bytes in 13 allocations in file 'taskprocessor.c' 4536 bytes in 21 allocations in file 'file.c' 1514 bytes in 8 allocations in file 'res_musiconhold.c' 23135 bytes in 297 allocations in file 'config.c' 10600 bytes in 46 allocations in file 'sched.c' 1832 bytes in 77 allocations in file 'manager.c' 96 bytes in 2 allocations in file 'devicestate.c' 44390 bytes in 527 allocations in file 'xmldoc.c' 33793 bytes in 361 allocations in file 'res_phoneprov.c' 2511 bytes in 34 allocations in file 'utils.c' 9584 bytes in 491 allocations in file 'cli.c' 39199 bytes in 1191 allocations in file 'indications.c' 14714 bytes in 486 allocations in file 'loader.c' 7821 bytes in 278 allocations in file 'asterisk.c' 341422 bytes in 1137 allocations in file 'pbx.c' 394740 bytes in 32895 allocations in file 'astobj2.c' 2077868 bytes in 32797 allocations in file 'chan_iax2.c' 3192135 bytes allocated in 70859 allocations *CLI> <--- SIP read from UDP:10.211.55.2:5060 ---> <-------------> creating audio RTP session <--- SIP read from UDP:10.211.55.2:5060 ---> INVITE sip:1002@10.211.55.3 SIP/2.0 Via: SIP/2.0/UDP 10.211.55.2:5060;rport;branch=z9hG4bKPjEED9200D-15BB-4AE6-A6AD-7180A35FE476 Max-Forwards: 70 From: sip:1001@10.211.55.3;tag=16E0B701-346C-4C89-BACB-7EE163A695C7 To: sip:1002@10.211.55.3 Contact: Call-ID: 19092D7E-FD0D-4943-8948-AF7D9DDB44D1 CSeq: 27696 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800 Min-SE: 90 User-Agent: PJSUA v1.5/i386-apple-darwin8.11.1 Content-Type: application/sdp Content-Length: 456 v=0 o=- 3473851031 3473851031 IN IP4 10.127.1.99 s=pjmedia c=IN IP4 10.127.1.99 t=0 0 a=X-nat:0 m=audio 4000 RTP/AVP 103 102 104 113 3 0 8 9 101 a=rtcp:4001 IN IP4 10.127.1.99 a=rtpmap:103 speex/16000 a=rtpmap:102 speex/8000 a=rtpmap:104 speex/32000 a=rtpmap:113 iLBC/8000 a=fmtp:113 mode=30 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=sendrecv a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> --- (15 headers 20 lines) --- [Jan 30 09:41:06] WARNING[22245]: rtp.c:2573 ast_rtp_new_with_bindaddr: rtp_new_with_bindaddr called 0x9062fe8 Sending to 10.211.55.2 : 5060 (no NAT) Using INVITE request as basis request - 19092D7E-FD0D-4943-8948-AF7D9DDB44D1 Found peer '1001' for '1001' from 10.211.55.2:5060 <--- Reliably Transmitting (no NAT) to 10.211.55.2:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.211.55.2:5060;branch=z9hG4bKPjEED9200D-15BB-4AE6-A6AD-7180A35FE476;received=10.211.55.2;rport=5060 From: sip:1001@10.211.55.3;tag=16E0B701-346C-4C89-BACB-7EE163A695C7 To: sip:1002@10.211.55.3;tag=as36499984 Call-ID: 19092D7E-FD0D-4943-8948-AF7D9DDB44D1 CSeq: 27696 INVITE Server: Asterisk PBX 1.6.2.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7fd498e6" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '19092D7E-FD0D-4943-8948-AF7D9DDB44D1' in 32000 ms (Method: INVITE) <--- SIP read from UDP:10.211.55.2:5060 ---> ACK sip:1002@10.211.55.3 SIP/2.0 Via: SIP/2.0/UDP 10.211.55.2:5060;rport;branch=z9hG4bKPjEED9200D-15BB-4AE6-A6AD-7180A35FE476 Max-Forwards: 70 From: sip:1001@10.211.55.3;tag=16E0B701-346C-4C89-BACB-7EE163A695C7 To: sip:1002@10.211.55.3;tag=as36499984 Call-ID: 19092D7E-FD0D-4943-8948-AF7D9DDB44D1 CSeq: 27696 ACK Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from UDP:10.211.55.2:5060 ---> INVITE sip:1002@10.211.55.3 SIP/2.0 Via: SIP/2.0/UDP 10.211.55.2:5060;rport;branch=z9hG4bKPjA6E10A6C-51C1-489B-AA69-9B83D4267537 Max-Forwards: 70 From: sip:1001@10.211.55.3;tag=16E0B701-346C-4C89-BACB-7EE163A695C7 To: sip:1002@10.211.55.3 Contact: Call-ID: 19092D7E-FD0D-4943-8948-AF7D9DDB44D1 CSeq: 27697 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800 Min-SE: 90 User-Agent: PJSUA v1.5/i386-apple-darwin8.11.1 Authorization: Digest username="1001", realm="asterisk", nonce="7fd498e6", uri="sip:1002@10.211.55.3", response="a220f7ab31771d5e2636697f0e3cd8d2", algorithm=MD5 Content-Type: application/sdp Content-Length: 456 v=0 o=- 3473851031 3473851031 IN IP4 10.127.1.99 s=pjmedia c=IN IP4 10.127.1.99 t=0 0 a=X-nat:0 m=audio 4000 RTP/AVP 103 102 104 113 3 0 8 9 101 a=rtcp:4001 IN IP4 10.127.1.99 a=rtpmap:103 speex/16000 a=rtpmap:102 speex/8000 a=rtpmap:104 speex/32000 a=rtpmap:113 iLBC/8000 a=fmtp:113 mode=30 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=sendrecv a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> --- (16 headers 20 lines) --- Sending to 10.211.55.2 : 5060 (no NAT) Using INVITE request as basis request - 19092D7E-FD0D-4943-8948-AF7D9DDB44D1 Found peer '1001' for '1001' from 10.211.55.2:5060 Found RTP audio format 103 Found RTP audio format 102 Found RTP audio format 104 Found RTP audio format 113 Found RTP audio format 3 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 9 Found RTP audio format 101 Found audio description format speex for ID 103 Found audio description format speex for ID 102 Found audio description format speex for ID 104 Found audio description format iLBC for ID 113 Found audio description format GSM for ID 3 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format G722 for ID 9 Found audio description format telephone-event for ID 101 Capabilities: us - 0x2 (gsm), peer - audio=0x50160e (gsm|ulaw|alaw|speex|ilbc|g722|h263p|mpeg4)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x2 (gsm) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.127.1.99:4000 Looking for 1002 in my-phones (domain 10.211.55.3) list_route: hop: <--- Transmitting (no NAT) to 10.211.55.2:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.211.55.2:5060;branch=z9hG4bKPjA6E10A6C-51C1-489B-AA69-9B83D4267537;received=10.211.55.2;rport=5060 From: sip:1001@10.211.55.3;tag=16E0B701-346C-4C89-BACB-7EE163A695C7 To: sip:1002@10.211.55.3 Call-ID: 19092D7E-FD0D-4943-8948-AF7D9DDB44D1 CSeq: 27697 INVITE Server: Asterisk PBX 1.6.2.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <------------> [New Thread -1228563552 (LWP 22266)] creating audio RTP session [Jan 30 09:41:06] WARNING[22266]: rtp.c:2573 ast_rtp_new_with_bindaddr: rtp_new_with_bindaddr called 0x9067b18 Really destroying SIP dialog '7662993008715b22774df9a56804d2de@127.0.1.1' Method: INVITE [Jan 30 09:41:06] WARNING[22266]: rtp.c:3140 ast_rtp_destroy: rtp_destory called 0x9067b18 [Jan 30 09:41:06] WARNING[22266]: app_dial.c:1745 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) <--- Reliably Transmitting (no NAT) to 10.211.55.2:5060 ---> SIP/2.0 503 Service Unavailable Via: SIP/2.0/UDP 10.211.55.2:5060;branch=z9hG4bKPjA6E10A6C-51C1-489B-AA69-9B83D4267537;received=10.211.55.2;rport=5060 From: sip:1001@10.211.55.3;tag=16E0B701-346C-4C89-BACB-7EE163A695C7 To: sip:1002@10.211.55.3;tag=as691f3a39 Call-ID: 19092D7E-FD0D-4943-8948-AF7D9DDB44D1 CSeq: 27697 INVITE Server: Asterisk PBX 1.6.2.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Content-Length: 0 X-Asterisk-HangupCause: Unknown X-Asterisk-HangupCauseCode: 20 <------------> <--- SIP read from UDP:10.211.55.2:5060 ---> ACK sip:1002@10.211.55.3 SIP/2.0 Via: SIP/2.0/UDP 10.211.55.2:5060;rport;branch=z9hG4bKPjA6E10A6C-51C1-489B-AA69-9B83D4267537 Max-Forwards: 70 From: sip:1001@10.211.55.3;tag=16E0B701-346C-4C89-BACB-7EE163A695C7 To: sip:1002@10.211.55.3;tag=as691f3a39 Call-ID: 19092D7E-FD0D-4943-8948-AF7D9DDB44D1 CSeq: 27697 ACK Content-Length: 0 <-------------> --- (8 headers 0 lines) --- [Thread -1228563552 (zombie) exited] <--- SIP read from UDP:10.211.55.2:5060 ---> <-------------> *CLI> Really destroying SIP dialog '9845EE6A-E8AE-4EF2-922D-372B845142E2' Method: REGISTER Really destroying SIP dialog '5ACEDF97-17FF-4028-A4C5-EDBE79D7BEED' Method: REGISTER memory show summary memory show summary 484 bytes in 1 allocations in file 'app_confbridge.c' 304 bytes in 2 allocations in file 'hashtab.c' 752 bytes in 2 allocations in file 'res_crypto.c' 1 bytes in 1 allocations in file 'app_dial.c' 936 bytes in 1 allocations in file 'ssl.c' 76 bytes in 1 allocations in file 'pbx_ael.c' 484 bytes in 1 allocations in file 'bridge_multiplexed.c' 244 bytes in 1 allocations in file 'iax2-provision.c' 14 bytes in 2 allocations in file 'http.c' 12324 bytes in 1 allocations in file 'stdtime/localtime.c' 548 bytes in 2 allocations in file 'features.c' 356 bytes in 1 allocations in file 'func_dialgroup.c' 484 bytes in 1 allocations in file 'app_queue.c' 1132 bytes in 7 allocations in file '/home/areyouwireless/asterisk/asterisk-1.6.2.1/include/asterisk/threadsto' 4096 bytes in 1 allocations in file 'chan_unistim.c' 1704 bytes in 2 allocations in file 'chan_oss.c' 216 bytes in 2 allocations in file 'cdr.c' 9312 bytes in 3 allocations in file 'mpool/mpool.c' 12100 bytes in 2 allocations in file 'rtp.c' 4644 bytes in 2 allocations in file 'res_timing_pthread.c' 576 bytes in 4 allocations in file 'logger.c' 25440 bytes in 2 allocations in file 'app_followme.c' 2119 bytes in 2 allocations in file '/home/areyouwireless/asterisk/asterisk-1.6.2.1/include/asterisk/strings.h' 212 bytes in 3 allocations in file 'netsock.c' 35963 bytes in 10 allocations in file 'app_minivm.c' 14138 bytes in 13 allocations in file 'event.c' 80 bytes in 10 allocations in file 'channel.c' 1395 bytes in 6 allocations in file 'res_clialiases.c' 24590 bytes in 17 allocations in file 'chan_sip.c' 84 bytes in 4 allocations in file 'timing.c' 969 bytes in 60 allocations in file 'ael/pval.c' 6980 bytes in 8 allocations in file 'app_voicemail.c' 30888 bytes in 24 allocations in file 'io.c' 779 bytes in 13 allocations in file 'taskprocessor.c' 4536 bytes in 21 allocations in file 'file.c' 1514 bytes in 8 allocations in file 'res_musiconhold.c' 23135 bytes in 297 allocations in file 'config.c' 10616 bytes in 46 allocations in file 'sched.c' 1832 bytes in 77 allocations in file 'manager.c' 96 bytes in 2 allocations in file 'devicestate.c' 44390 bytes in 527 allocations in file 'xmldoc.c' 33793 bytes in 361 allocations in file 'res_phoneprov.c' 2511 bytes in 34 allocations in file 'utils.c' 9584 bytes in 491 allocations in file 'cli.c' 39199 bytes in 1191 allocations in file 'indications.c' 14714 bytes in 486 allocations in file 'loader.c' 7821 bytes in 278 allocations in file 'asterisk.c' 341572 bytes in 1147 allocations in file 'pbx.c' 394716 bytes in 32893 allocations in file 'astobj2.c' 2077868 bytes in 32797 allocations in file 'chan_iax2.c' 3202321 bytes allocated in 70868 allocations *CLI> <--- SIP read from UDP:10.211.55.2:5060 ---> <-------------> creating audio RTP session <--- SIP read from UDP:10.211.55.2:5060 ---> INVITE sip:1002@10.211.55.3 SIP/2.0 Via: SIP/2.0/UDP 10.211.55.2:5060;rport;branch=z9hG4bKPj9A57BAED-1253-4F41-9760-AA1EA3E71A2B Max-Forwards: 70 From: sip:1001@10.211.55.3;tag=EB62B781-2A3E-478B-AD58-18333E9FA5E6 To: sip:1002@10.211.55.3 Contact: Call-ID: 3E951DFB-47B7-4016-9C39-1EBF279A456A CSeq: 12140 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800 Min-SE: 90 User-Agent: PJSUA v1.5/i386-apple-darwin8.11.1 Content-Type: application/sdp Content-Length: 456 v=0 o=- 3473851059 3473851059 IN IP4 10.127.1.99 s=pjmedia c=IN IP4 10.127.1.99 t=0 0 a=X-nat:0 m=audio 4002 RTP/AVP 103 102 104 113 3 0 8 9 101 a=rtcp:4003 IN IP4 10.127.1.99 a=rtpmap:103 speex/16000 a=rtpmap:102 speex/8000 a=rtpmap:104 speex/32000 a=rtpmap:113 iLBC/8000 a=fmtp:113 mode=30 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=sendrecv a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> --- (15 headers 20 lines) --- [Jan 30 09:41:34] WARNING[22245]: rtp.c:2573 ast_rtp_new_with_bindaddr: rtp_new_with_bindaddr called 0x9068920 Sending to 10.211.55.2 : 5060 (no NAT) Using INVITE request as basis request - 3E951DFB-47B7-4016-9C39-1EBF279A456A Found peer '1001' for '1001' from 10.211.55.2:5060 <--- Reliably Transmitting (no NAT) to 10.211.55.2:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.211.55.2:5060;branch=z9hG4bKPj9A57BAED-1253-4F41-9760-AA1EA3E71A2B;received=10.211.55.2;rport=5060 From: sip:1001@10.211.55.3;tag=EB62B781-2A3E-478B-AD58-18333E9FA5E6 To: sip:1002@10.211.55.3;tag=as08219649 Call-ID: 3E951DFB-47B7-4016-9C39-1EBF279A456A CSeq: 12140 INVITE Server: Asterisk PBX 1.6.2.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4949e420" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '3E951DFB-47B7-4016-9C39-1EBF279A456A' in 32000 ms (Method: INVITE) <--- SIP read from UDP:10.211.55.2:5060 ---> ACK sip:1002@10.211.55.3 SIP/2.0 Via: SIP/2.0/UDP 10.211.55.2:5060;rport;branch=z9hG4bKPj9A57BAED-1253-4F41-9760-AA1EA3E71A2B Max-Forwards: 70 From: sip:1001@10.211.55.3;tag=EB62B781-2A3E-478B-AD58-18333E9FA5E6 To: sip:1002@10.211.55.3;tag=as08219649 Call-ID: 3E951DFB-47B7-4016-9C39-1EBF279A456A CSeq: 12140 ACK Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from UDP:10.211.55.2:5060 ---> INVITE sip:1002@10.211.55.3 SIP/2.0 Via: SIP/2.0/UDP 10.211.55.2:5060;rport;branch=z9hG4bKPjACDA848A-8C05-4748-ADF3-6CD9189798D1 Max-Forwards: 70 From: sip:1001@10.211.55.3;tag=EB62B781-2A3E-478B-AD58-18333E9FA5E6 To: sip:1002@10.211.55.3 Contact: Call-ID: 3E951DFB-47B7-4016-9C39-1EBF279A456A CSeq: 12141 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800 Min-SE: 90 User-Agent: PJSUA v1.5/i386-apple-darwin8.11.1 Authorization: Digest username="1001", realm="asterisk", nonce="4949e420", uri="sip:1002@10.211.55.3", response="401fbe6903c051904904e27e3c3e4e5b", algorithm=MD5 Content-Type: application/sdp Content-Length: 456 v=0 o=- 3473851059 3473851059 IN IP4 10.127.1.99 s=pjmedia c=IN IP4 10.127.1.99 t=0 0 a=X-nat:0 m=audio 4002 RTP/AVP 103 102 104 113 3 0 8 9 101 a=rtcp:4003 IN IP4 10.127.1.99 a=rtpmap:103 speex/16000 a=rtpmap:102 speex/8000 a=rtpmap:104 speex/32000 a=rtpmap:113 iLBC/8000 a=fmtp:113 mode=30 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=sendrecv a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> --- (16 headers 20 lines) --- Sending to 10.211.55.2 : 5060 (no NAT) Using INVITE request as basis request - 3E951DFB-47B7-4016-9C39-1EBF279A456A Found peer '1001' for '1001' from 10.211.55.2:5060 Found RTP audio format 103 Found RTP audio format 102 Found RTP audio format 104 Found RTP audio format 113 Found RTP audio format 3 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 9 Found RTP audio format 101 Found audio description format speex for ID 103 Found audio description format speex for ID 102 Found audio description format speex for ID 104 Found audio description format iLBC for ID 113 Found audio description format GSM for ID 3 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format G722 for ID 9 Found audio description format telephone-event for ID 101 Capabilities: us - 0x2 (gsm), peer - audio=0x50160e (gsm|ulaw|alaw|speex|ilbc|g722|h263p|mpeg4)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x2 (gsm) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.127.1.99:4002 Looking for 1002 in my-phones (domain 10.211.55.3) list_route: hop: <--- Transmitting (no NAT) to 10.211.55.2:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.211.55.2:5060;branch=z9hG4bKPjACDA848A-8C05-4748-ADF3-6CD9189798D1;received=10.211.55.2;rport=5060 From: sip:1001@10.211.55.3;tag=EB62B781-2A3E-478B-AD58-18333E9FA5E6 To: sip:1002@10.211.55.3 Call-ID: 3E951DFB-47B7-4016-9C39-1EBF279A456A CSeq: 12141 INVITE Server: Asterisk PBX 1.6.2.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <------------> [New Thread -1228563552 (LWP 22267)] creating audio RTP session [Jan 30 09:41:34] WARNING[22267]: rtp.c:2573 ast_rtp_new_with_bindaddr: rtp_new_with_bindaddr called 0x906ed90 Really destroying SIP dialog '615e9e602d2ae388081832c1490c5f82@127.0.1.1' Method: INVITE [Jan 30 09:41:34] WARNING[22267]: rtp.c:3140 ast_rtp_destroy: rtp_destory called 0x906ed90 [Jan 30 09:41:34] WARNING[22267]: app_dial.c:1745 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) <--- Reliably Transmitting (no NAT) to 10.211.55.2:5060 ---> SIP/2.0 503 Service Unavailable Via: SIP/2.0/UDP 10.211.55.2:5060;branch=z9hG4bKPjACDA848A-8C05-4748-ADF3-6CD9189798D1;received=10.211.55.2;rport=5060 From: sip:1001@10.211.55.3;tag=EB62B781-2A3E-478B-AD58-18333E9FA5E6 To: sip:1002@10.211.55.3;tag=as3fefd9f3 Call-ID: 3E951DFB-47B7-4016-9C39-1EBF279A456A CSeq: 12141 INVITE Server: Asterisk PBX 1.6.2.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Content-Length: 0 X-Asterisk-HangupCause: Unknown X-Asterisk-HangupCauseCode: 20 <------------> <--- SIP read from UDP:10.211.55.2:5060 ---> ACK sip:1002@10.211.55.3 SIP/2.0 Via: SIP/2.0/UDP 10.211.55.2:5060;rport;branch=z9hG4bKPjACDA848A-8C05-4748-ADF3-6CD9189798D1 Max-Forwards: 70 From: sip:1001@10.211.55.3;tag=EB62B781-2A3E-478B-AD58-18333E9FA5E6 To: sip:1002@10.211.55.3;tag=as3fefd9f3 Call-ID: 3E951DFB-47B7-4016-9C39-1EBF279A456A CSeq: 12141 ACK Content-Length: 0 <-------------> --- (8 headers 0 lines) --- [Thread -1228563552 (zombie) exited] *CLI> <--- SIP read from UDP:10.211.55.2:5060 ---> <-------------> memory show summary memory show summary 484 bytes in 1 allocations in file 'app_confbridge.c' 304 bytes in 2 allocations in file 'hashtab.c' 752 bytes in 2 allocations in file 'res_crypto.c' 1 bytes in 1 allocations in file 'app_dial.c' 936 bytes in 1 allocations in file 'ssl.c' 76 bytes in 1 allocations in file 'pbx_ael.c' 484 bytes in 1 allocations in file 'bridge_multiplexed.c' 244 bytes in 1 allocations in file 'iax2-provision.c' 14 bytes in 2 allocations in file 'http.c' 12324 bytes in 1 allocations in file 'stdtime/localtime.c' 548 bytes in 2 allocations in file 'features.c' 356 bytes in 1 allocations in file 'func_dialgroup.c' 484 bytes in 1 allocations in file 'app_queue.c' 1132 bytes in 7 allocations in file '/home/areyouwireless/asterisk/asterisk-1.6.2.1/include/asterisk/threadsto' 4096 bytes in 1 allocations in file 'chan_unistim.c' 1704 bytes in 2 allocations in file 'chan_oss.c' 216 bytes in 2 allocations in file 'cdr.c' 9312 bytes in 3 allocations in file 'mpool/mpool.c' 24200 bytes in 4 allocations in file 'rtp.c' 4644 bytes in 2 allocations in file 'res_timing_pthread.c' 576 bytes in 4 allocations in file 'logger.c' 25440 bytes in 2 allocations in file 'app_followme.c' 2119 bytes in 2 allocations in file '/home/areyouwireless/asterisk/asterisk-1.6.2.1/include/asterisk/strings.h' 212 bytes in 3 allocations in file 'netsock.c' 35963 bytes in 10 allocations in file 'app_minivm.c' 14138 bytes in 13 allocations in file 'event.c' 80 bytes in 10 allocations in file 'channel.c' 1395 bytes in 6 allocations in file 'res_clialiases.c' 28772 bytes in 22 allocations in file 'chan_sip.c' 84 bytes in 4 allocations in file 'timing.c' 969 bytes in 60 allocations in file 'ael/pval.c' 6980 bytes in 8 allocations in file 'app_voicemail.c' 30888 bytes in 24 allocations in file 'io.c' 779 bytes in 13 allocations in file 'taskprocessor.c' 4536 bytes in 21 allocations in file 'file.c' 1514 bytes in 8 allocations in file 'res_musiconhold.c' 23135 bytes in 297 allocations in file 'config.c' 10636 bytes in 47 allocations in file 'sched.c' 1832 bytes in 77 allocations in file 'manager.c' 96 bytes in 2 allocations in file 'devicestate.c' 44390 bytes in 527 allocations in file 'xmldoc.c' 33793 bytes in 361 allocations in file 'res_phoneprov.c' 2511 bytes in 34 allocations in file 'utils.c' 9584 bytes in 491 allocations in file 'cli.c' 39199 bytes in 1191 allocations in file 'indications.c' 14714 bytes in 486 allocations in file 'loader.c' 7821 bytes in 278 allocations in file 'asterisk.c' 341572 bytes in 1147 allocations in file 'pbx.c' 394716 bytes in 32893 allocations in file 'astobj2.c' 2077868 bytes in 32797 allocations in file 'chan_iax2.c' 3218623 bytes allocated in 70876 allocations *CLI> <--- SIP read from UDP:10.211.55.2:5060 ---> REGISTER sip:10.211.55.3 SIP/2.0 Via: SIP/2.0/UDP 10.211.55.2:5060;rport;branch=z9hG4bKPj4F50EDCB-E2F4-4329-A5C6-54C71FB779B8 Max-Forwards: 70 From: ;tag=393F4635-5B32-40D2-8661-B994961168C3 To: Call-ID: 5ACEDF97-17FF-4028-A4C5-EDBE79D7BEED CSeq: 15063 REGISTER User-Agent: PJSUA v1.5/i386-apple-darwin8.11.1 Contact: Expires: 0 Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Sending to 10.211.55.2 : 5060 (no NAT) <--- Transmitting (no NAT) to 10.211.55.2:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.211.55.2:5060;branch=z9hG4bKPj4F50EDCB-E2F4-4329-A5C6-54C71FB779B8;received=10.211.55.2;rport=5060 From: ;tag=393F4635-5B32-40D2-8661-B994961168C3 To: ;tag=as426dec76 Call-ID: 5ACEDF97-17FF-4028-A4C5-EDBE79D7BEED CSeq: 15063 REGISTER Server: Asterisk PBX 1.6.2.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="45b507d3" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '5ACEDF97-17FF-4028-A4C5-EDBE79D7BEED' in 32000 ms (Method: REGISTER) <--- SIP read from UDP:10.211.55.2:5060 ---> REGISTER sip:10.211.55.3 SIP/2.0 Via: SIP/2.0/UDP 10.211.55.2:5060;rport;branch=z9hG4bKPj011B535F-1B15-4994-9412-EF10C7F7783B Max-Forwards: 70 From: ;tag=393F4635-5B32-40D2-8661-B994961168C3 To: Call-ID: 5ACEDF97-17FF-4028-A4C5-EDBE79D7BEED CSeq: 15064 REGISTER User-Agent: PJSUA v1.5/i386-apple-darwin8.11.1 Contact: Expires: 0 Authorization: Digest username="1001", realm="asterisk", nonce="45b507d3", uri="sip:10.211.55.3", response="4ad6959feb36a5c8babf921ecfff0d22", algorithm=MD5 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Sending to 10.211.55.2 : 5060 (no NAT) <--- Transmitting (no NAT) to 10.211.55.2:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.211.55.2:5060;branch=z9hG4bKPj011B535F-1B15-4994-9412-EF10C7F7783B;received=10.211.55.2;rport=5060 From: ;tag=393F4635-5B32-40D2-8661-B994961168C3 To: ;tag=as426dec76 Call-ID: 5ACEDF97-17FF-4028-A4C5-EDBE79D7BEED CSeq: 15064 REGISTER Server: Asterisk PBX 1.6.2.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Expires: 0 Date: Sat, 30 Jan 2010 14:41:51 GMT Content-Length: 0 <------------> Scheduling destruction of SIP dialog '5ACEDF97-17FF-4028-A4C5-EDBE79D7BEED' in 32000 ms (Method: REGISTER) *CLI> memory show summary memory show summary 484 bytes in 1 allocations in file 'app_confbridge.c' 304 bytes in 2 allocations in file 'hashtab.c' 752 bytes in 2 allocations in file 'res_crypto.c' 1 bytes in 1 allocations in file 'app_dial.c' 936 bytes in 1 allocations in file 'ssl.c' 76 bytes in 1 allocations in file 'pbx_ael.c' 484 bytes in 1 allocations in file 'bridge_multiplexed.c' 244 bytes in 1 allocations in file 'iax2-provision.c' 14 bytes in 2 allocations in file 'http.c' 12324 bytes in 1 allocations in file 'stdtime/localtime.c' 548 bytes in 2 allocations in file 'features.c' 356 bytes in 1 allocations in file 'func_dialgroup.c' 484 bytes in 1 allocations in file 'app_queue.c' 1132 bytes in 7 allocations in file '/home/areyouwireless/asterisk/asterisk-1.6.2.1/include/asterisk/threadsto' 4096 bytes in 1 allocations in file 'chan_unistim.c' 1704 bytes in 2 allocations in file 'chan_oss.c' 216 bytes in 2 allocations in file 'cdr.c' 9312 bytes in 3 allocations in file 'mpool/mpool.c' 24200 bytes in 4 allocations in file 'rtp.c' 4644 bytes in 2 allocations in file 'res_timing_pthread.c' 576 bytes in 4 allocations in file 'logger.c' 25440 bytes in 2 allocations in file 'app_followme.c' 2119 bytes in 2 allocations in file '/home/areyouwireless/asterisk/asterisk-1.6.2.1/include/asterisk/strings.h' 212 bytes in 3 allocations in file 'netsock.c' 35963 bytes in 10 allocations in file 'app_minivm.c' 14138 bytes in 13 allocations in file 'event.c' 80 bytes in 10 allocations in file 'channel.c' 1395 bytes in 6 allocations in file 'res_clialiases.c' 84 bytes in 4 allocations in file 'timing.c' 969 bytes in 60 allocations in file 'ael/pval.c' 6980 bytes in 8 allocations in file 'app_voicemail.c' 30888 bytes in 24 allocations in file 'io.c' 32211 bytes in 25 allocations in file 'chan_sip.c' 779 bytes in 13 allocations in file 'taskprocessor.c' 4536 bytes in 21 allocations in file 'file.c' 1514 bytes in 8 allocations in file 'res_musiconhold.c' 23135 bytes in 297 allocations in file 'config.c' 1832 bytes in 77 allocations in file 'manager.c' 96 bytes in 2 allocations in file 'devicestate.c' 44390 bytes in 527 allocations in file 'xmldoc.c' 33793 bytes in 361 allocations in file 'res_phoneprov.c' 2511 bytes in 34 allocations in file 'utils.c' 9585 bytes in 491 allocations in file 'cli.c' 39199 bytes in 1191 allocations in file 'indications.c' 14714 bytes in 486 allocations in file 'loader.c' 7821 bytes in 278 allocations in file 'asterisk.c' 341572 bytes in 1147 allocations in file 'pbx.c' 394728 bytes in 32894 allocations in file 'astobj2.c' 2077868 bytes in 32797 allocations in file 'chan_iax2.c' 10656 bytes in 48 allocations in file 'sched.c' 3222095 bytes allocated in 70881 allocations *CLI> Program received signal SIGINT, Interrupt. [Switching to Thread -1213818432 (LWP 22227)] 0xffffe410 in __kernel_vsyscall () (gdb) kill Kill the program being debugged? (y or n) y (gdb) quit Debugger finished