[root@atlsvrmus01 asterisk-1.4]# asterisk -r Asterisk SVN-branch-1.4-r61781, Copyright (C) 1999 - 2006 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. ========================================================================= Connected to Asterisk SVN-branch-1.4-r61781 currently running on atlsvrmus01 (pid = 16760) onnection01*CLI> Verbosity is at least 100 atlsvrmus01*CLI> core set debug 100 Core debug is at least 100 atlsvrmus01*CLI> core set verbose 100 Verbosity is at least 100 atlsvrmus01*CLI> core set debug channel all Debugging on new channels is enabled atlsvrmus01*CLI> <--- SIP read from 192.168.108.101:5060 ---> INVITE sip:907303@192.168.100.18:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.108.101:5060;branch=z9hG4bK410a3b05 Remote-Party-ID: ;party=calling;screen=yes;privacy=off From: ;tag=d1864fa0-11e4-4647-b0cf-069f6a99d47f-42000140 To: Date: Tue, 24 Apr 2007 20:51:09 GMT Call-ID: 8604e200-62e16dbd-39b-656ca8c0@192.168.108.101 Supported: timer,replaces Min-SE: 1800 User-Agent: Cisco-CCM5.0 Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY CSeq: 101 INVITE Contact: Expires: 180 Allow-Events: presence Session-Expires: 1800 Max-Forwards: 70 Content-Length: 0 <-------------> --- (18 headers 0 lines) --- Sending to 192.168.108.101 : 5060 (no NAT) Using INVITE request as basis request - 8604e200-62e16dbd-39b-656ca8c0@192.168.108.101 Found no matching peer or user for '192.168.108.101:5060' Looking for 907303 in default (domain 192.168.100.18) list_route: hop: <--- Transmitting (no NAT) to 192.168.108.101:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.108.101:5060;branch=z9hG4bK410a3b05;received=192.168.108.101 From: ;tag=d1864fa0-11e4-4647-b0cf-069f6a99d47f-42000140 To: Call-ID: 8604e200-62e16dbd-39b-656ca8c0@192.168.108.101 CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> -- Executing [907303@default:1] MeetMe("SIP/192.168.108.101-09fe4378", "303|AwxdPMcsr|147369") in new stack Audio is at 192.168.100.18 port 17862 Adding codec 0x2 (gsm) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP atlsvrmus01*CLI> <--- Reliably Transmitting (no NAT) to 192.168.108.101:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.108.101:5060;branch=z9hG4bK410a3b05;received=192.168.108.101 From: ;tag=d1864fa0-11e4-4647-b0cf-069f6a99d47f-42000140 To: ;tag=as14dd5c97 Call-ID: 8604e200-62e16dbd-39b-656ca8c0@192.168.108.101 CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 235 v=0 o=root 16760 16760 IN IP4 192.168.100.18 s=session c=IN IP4 192.168.100.18 t=0 0 m=audio 17862 RTP/AVP 3 0 8 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> -- Created MeetMe conference 1023 for conference '303' -- Playing 'conf-getpin' (language 'en') atlsvrmus01*CLI> <--- SIP read from 192.168.108.101:5060 ---> ACK sip:907303@192.168.100.18:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.108.101:5060;branch=z9hG4bKa6d5625 From: ;tag=d1864fa0-11e4-4647-b0cf-069f6a99d47f-42000140 To: ;tag=as14dd5c97 Date: Tue, 24 Apr 2007 20:51:09 GMT Call-ID: 8604e200-62e16dbd-39b-656ca8c0@192.168.108.101 Max-Forwards: 70 CSeq: 101 ACK Allow-Events: presence Content-Type: application/sdp Content-Length: 162 v=0 o=CiscoSystemsCCM-SIP 2000 1 IN IP4 192.168.108.101 s=SIP Call c=IN IP4 192.168.100.205 t=0 0 m=audio 28242 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=ptime:20 <-------------> --- (11 headers 8 lines) --- Found RTP audio format 0 Peer audio RTP is at port 192.168.100.205:28242 Found description format PCMU for ID 0 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 192.168.100.205:28242 << [ TYPE: Control (4) SUBCLASS: Unknown control '17' (17) ] [SIP/192.168.108.101-09fe4378] << [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/192.168.108.101-09fe4378] -- Playing 'conf-invalidpin' (language 'en') -- Playing 'conf-getpin' (language 'en') -- Playing 'conf-invalidpin' (language 'en') -- Playing 'conf-getpin' (language 'en') -- Playing 'conf-invalidpin' (language 'en') [Apr 24 15:54:19] WARNING[16873]: cdr_csv.c:265 writefile: Account code 'Zap/pseudo-63660427' insecure for writing file [Apr 24 15:54:19] WARNING[16873]: cdr_csv.c:306 csv_log: Unable to write CSV record to account file 'Zap/pseudo-63660427' : No such file or directory -- Hungup '**Unknown**' == Spawn extension (default, 907303, 1) exited non-zero on 'SIP/192.168.108.101-09fe4378' Scheduling destruction of SIP dialog '8604e200-62e16dbd-39b-656ca8c0@192.168.108.101' in 32000 ms (Method: ACK) set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.108.101, port 5060 Reliably Transmitting (no NAT) to 192.168.108.101:5060: BYE sip:5147@192.168.108.101:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.18:5060;branch=z9hG4bK30afbf2b;rport From: ;tag=as14dd5c97 To: ;tag=d1864fa0-11e4-4647-b0cf-069f6a99d47f-42000140 Call-ID: 8604e200-62e16dbd-39b-656ca8c0@192.168.108.101 CSeq: 102 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- atlsvrmus01*CLI> <--- SIP read from 192.168.108.101:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.100.18:5060;branch=z9hG4bK30afbf2b;rport From: ;tag=as14dd5c97 To: ;tag=d1864fa0-11e4-4647-b0cf-069f6a99d47f-42000140 Date: Tue, 24 Apr 2007 20:51:55 GMT Call-ID: 8604e200-62e16dbd-39b-656ca8c0@192.168.108.101 CSeq: 102 BYE Content-Length: 0 <-------------> --- (8 headers 0 lines) --- SIP Response message for INCOMING dialog BYE arrived Really destroying SIP dialog '8604e200-62e16dbd-39b-656ca8c0@192.168.108.101' Method: ACK atlsvrmus01*CLI>