[Apr 11 10:38:17] Asterisk SVN-branch-1.4-r60137, Copyright (C) 1999 - 2006 Digium, Inc. and others. [Apr 11 10:38:17] Created by Mark Spencer [Apr 11 10:38:17] Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. [Apr 11 10:38:17] This is free software, with components licensed under the GNU General Public [Apr 11 10:38:17] License version 2 and other licenses; you are welcome to redistribute it under [Apr 11 10:38:17] certain conditions. Type 'core show license' for details. [Apr 11 10:38:17] ========================================================================= [Apr 11 10:38:17] == Parsing '/etc/asterisk/asterisk.conf': Parsing /etc/asterisk/asterisk.conf [Apr 11 10:38:17] Found [Apr 11 10:38:17] == Parsing '/etc/asterisk/extconfig.conf': Parsing /etc/asterisk/extconfig.conf [Apr 11 10:38:17] Found [Apr 11 10:38:17] == Binding sipusers to odbc/mysql_config/sip [Apr 11 10:38:17] == Binding sippeers to odbc/mysql_config/sip [Apr 11 10:38:17] == Binding queues to odbc/mysql_config/queues [Apr 11 10:38:17] == Binding queue_members to odbc/mysql_config/queue_members [Apr 11 10:38:17] Asterisk Event Logger Started /var/log/asterisk/event_log [Apr 11 10:38:17] Asterisk Dynamic Loader Starting: [Apr 11 10:38:17] == Parsing '/etc/asterisk/modules.conf': [Apr 11 10:38:17] Found [Apr 11 10:38:17] Asterisk Ready. ]1;Asterisk]2;Asterisk Console on 'CapPBX' (pid 21295)*CLI> core set debug 45 Core debug was 5 and is now 4 *CLI> core set debuverbose 4 Verbosity was 5 and is now 4 *CLI> sip set debut No such command 'sip set debut' (type 'help' for help) *CLI> sip set debutg SIP Debugging enabled *CLI> [Apr 11 10:39:14] <--- SIP read from 192.168.2.41:5060 ---> INVITE sip:9XXXXXXXXXX@192.168.2.150:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.2.41;branch=z9hG4bK68e74187767BA3 From: "2400" ;tag=8FFD36C6-24E79537 To: CSeq: 1 INVITE Call-ID: 1132e582-c34ae83c-7628e665@192.168.2.41 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.1.0.2708 Supported: 100rel,replaces Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 249 v=0 o=- 1176301939 1176301939 IN IP4 192.168.2.41 s=Polycom IP Phone c=IN IP4 192.168.2.41 t=0 0 m=audio 2236 RTP/AVP 0 8 18 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 <-------------> [Apr 11 10:39:14] --- (14 headers 11 lines) --- [Apr 11 10:39:14] Sending to 192.168.2.41 : 5060 (no NAT) [Apr 11 10:39:14] Using INVITE request as basis request - 1132e582-c34ae83c-7628e665@192.168.2.41 [Apr 11 10:39:14] <--- Reliably Transmitting (no NAT) to 192.168.2.41:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.2.41;branch=z9hG4bK68e74187767BA3;received=192.168.2.41 From: "2400" ;tag=8FFD36C6-24E79537 To: ;tag=as551b4f2c Call-ID: 1132e582-c34ae83c-7628e665@192.168.2.41 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7445a374" Content-Length: 0 <------------> [Apr 11 10:39:14] Scheduling destruction of SIP dialog '1132e582-c34ae83c-7628e665@192.168.2.41' in 32000 ms (Method: INVITE) [Apr 11 10:39:14] Found user '2400' [Apr 11 10:39:14] <--- SIP read from 192.168.2.41:5060 ---> ACK sip:9XXXXXXXXXX@192.168.2.150:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.41;branch=z9hG4bK68e74187767BA3 From: "2400" ;tag=8FFD36C6-24E79537 To: ;tag=as551b4f2c CSeq: 1 ACK Call-ID: 1132e582-c34ae83c-7628e665@192.168.2.41 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.1.0.2708 Max-Forwards: 70 Content-Length: 0 <-------------> [Apr 11 10:39:14] --- (11 headers 0 lines) --- [Apr 11 10:39:14] <--- SIP read from 192.168.2.41:5060 ---> INVITE sip:9XXXXXXXXXX@192.168.2.150:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.2.41;branch=z9hG4bKa7be1899CE47DFA0 From: "2400" ;tag=8FFD36C6-24E79537 To: CSeq: 2 INVITE Call-ID: 1132e582-c34ae83c-7628e665@192.168.2.41 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.1.0.2708 Supported: 100rel,replaces Allow-Events: talk,hold,conference Proxy-Authorization: Digest username="2400", realm="asterisk", nonce="7445a374", uri="sip:9XXXXXXXXXX@192.168.2.150:5060;user=phone", response="6459a5119f1001fa41aefca3bcdc40e7", algorithm=MD5 Max-Forwards: 70 Content-Type: application/sdp Content-Length: 249 v=0 o=- 1176301939 1176301939 IN IP4 192.168.2.41 s=Polycom IP Phone c=IN IP4 192.168.2.41 t=0 0 m=audio 2236 RTP/AVP 0 8 18 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 <-------------> [Apr 11 10:39:14] --- (15 headers 11 lines) --- [Apr 11 10:39:14] Sending to 192.168.2.41 : 5060 (no NAT) [Apr 11 10:39:14] Using INVITE request as basis request - 1132e582-c34ae83c-7628e665@192.168.2.41 [Apr 11 10:39:14] Found user '2400' [Apr 11 10:39:14] Found RTP audio format 0 [Apr 11 10:39:14] Found RTP audio format 8 [Apr 11 10:39:14] Found RTP audio format 18 [Apr 11 10:39:14] Found RTP audio format 101 [Apr 11 10:39:14] Peer audio RTP is at port 192.168.2.41:2236 [Apr 11 10:39:14] Found description format PCMU for ID 0 [Apr 11 10:39:14] Found description format PCMA for ID 8 [Apr 11 10:39:14] Found description format G729 for ID 18 [Apr 11 10:39:14] Found description format telephone-event for ID 101 [Apr 11 10:39:14] Capabilities: us - 0x4 (ulaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x4 (ulaw) [Apr 11 10:39:14] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Apr 11 10:39:14] Peer audio RTP is at port 192.168.2.41:2236 [Apr 11 10:39:14] Looking for 9XXXXXXXXXX in sip-internal (domain 192.168.2.150) [Apr 11 10:39:14] list_route: hop: [Apr 11 10:39:14] <--- Transmitting (no NAT) to 192.168.2.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.2.41;branch=z9hG4bKa7be1899CE47DFA0;received=192.168.2.41 From: "2400" ;tag=8FFD36C6-24E79537 To: Call-ID: 1132e582-c34ae83c-7628e665@192.168.2.41 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Apr 11 10:39:14] -- Executing [9XXXXXXXXXX@sip-internal:1] Set("SIP/2400-08228e58", "SPYGROUP=10001") in new stack [Apr 11 10:39:14] -- Executing [9XXXXXXXXXX@sip-internal:2] Dial("SIP/2400-08228e58", "Zap/g1/9XXXXXXXXXX") in new stack [Apr 11 10:39:14] WARNING[21299]: chan_sip.c:16139 build_peer: Ignoring unknown option 'id' at line 0 of sip.conf! [Apr 11 10:39:14] WARNING[21299]: chan_sip.c:16139 build_peer: Ignoring unknown option 'name' at line 0 of sip.conf! [Apr 11 10:39:14] WARNING[21299]: chan_sip.c:16139 build_peer: Ignoring unknown option 'fullcontact' at line 0 of sip.conf! [Apr 11 10:39:14] -- Called g1/9XXXXXXXXXX [Apr 11 10:39:14] WARNING[21299]: chan_sip.c:16139 build_peer: Ignoring unknown option 'regseconds' at line 0 of sip.conf! [Apr 11 10:39:14] WARNING[21299]: chan_sip.c:16139 build_peer: Ignoring unknown option 'ipaddr' at line 0 of sip.conf! [Apr 11 10:39:14] WARNING[21299]: chan_sip.c:16139 build_peer: Ignoring unknown option 'cancallforward' at line 0 of sip.conf! [Apr 11 10:39:15] WARNING[21300]: app_voicemail.c:2299 inboxcount: Failed to obtain database object for ''! [Apr 11 10:39:15] Scheduling destruction of SIP dialog '31b7af1e22dacde605da8c9f504963ec@192.168.2.150' in 32000 ms (Method: NOTIFY) [Apr 11 10:39:15] Reliably Transmitting (no NAT) to 192.168.2.41:5060: NOTIFY sip:2400@192.168.2.41 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.150:5060;branch=z9hG4bK4316bdb9;rport From: "asterisk" ;tag=as75107102 To: Contact: Call-ID: 31b7af1e22dacde605da8c9f504963ec@192.168.2.150 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 93 Messages-Waiting: no Message-Account: sip:asterisk@192.168.2.150 Voice-Message: 0/0 (0/0) --- [Apr 11 10:39:15] <--- SIP read from 192.168.2.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.150:5060;branch=z9hG4bK4316bdb9;rport From: "asterisk" ;tag=as75107102 To: ;tag=40763F1F-F70FDE44 CSeq: 102 NOTIFY Call-ID: 31b7af1e22dacde605da8c9f504963ec@192.168.2.150 Contact: Event: message-summary User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.1.0.2708 Content-Length: 0 <-------------> [Apr 11 10:39:15] --- (10 headers 0 lines) --- [Apr 11 10:39:15] Really destroying SIP dialog '31b7af1e22dacde605da8c9f504963ec@192.168.2.150' Method: NOTIFY [Apr 11 10:39:18] Audio is at 192.168.2.150 port 14900 [Apr 11 10:39:18] Adding codec 0x4 (ulaw) to SDP [Apr 11 10:39:18] Adding non-codec 0x1 (telephone-event) to SDP [Apr 11 10:39:18] <--- Transmitting (no NAT) to 192.168.2.41:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.2.41;branch=z9hG4bKa7be1899CE47DFA0;received=192.168.2.41 From: "2400" ;tag=8FFD36C6-24E79537 To: ;tag=as0fd29470 Call-ID: 1132e582-c34ae83c-7628e665@192.168.2.41 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 242 v=0 o=root 21295 21295 IN IP4 192.168.2.150 s=session c=IN IP4 192.168.2.150 t=0 0 m=audio 14900 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Apr 11 10:39:18] -- Zap/1-1 is ringing [Apr 11 10:39:18] <--- Transmitting (no NAT) to 192.168.2.41:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.2.41;branch=z9hG4bKa7be1899CE47DFA0;received=192.168.2.41 From: "2400" ;tag=8FFD36C6-24E79537 To: ;tag=as0fd29470 Call-ID: 1132e582-c34ae83c-7628e665@192.168.2.41 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Apr 11 10:39:25] -- Zap/1-1 answered SIP/2400-08228e58 [Apr 11 10:39:25] Audio is at 192.168.2.150 port 14900 [Apr 11 10:39:25] Adding codec 0x4 (ulaw) to SDP [Apr 11 10:39:25] Adding non-codec 0x1 (telephone-event) to SDP [Apr 11 10:39:25] <--- Reliably Transmitting (no NAT) to 192.168.2.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.41;branch=z9hG4bKa7be1899CE47DFA0;received=192.168.2.41 From: "2400" ;tag=8FFD36C6-24E79537 To: ;tag=as0fd29470 Call-ID: 1132e582-c34ae83c-7628e665@192.168.2.41 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 242 v=0 o=root 21295 21296 IN IP4 192.168.2.150 s=session c=IN IP4 192.168.2.150 t=0 0 m=audio 14900 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Apr 11 10:39:25] WARNING[21319]: cdr.c:482 ast_cdr_merge: CDR start disagreement for SIP/2400-08228e58 [Apr 11 10:39:25] <--- SIP read from 192.168.2.41:5060 ---> ACK sip:9XXXXXXXXXX@192.168.2.150 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.41;branch=z9hG4bKfca950e86B3E9C1 From: "2400" ;tag=8FFD36C6-24E79537 To: ;tag=as0fd29470 CSeq: 2 ACK Call-ID: 1132e582-c34ae83c-7628e665@192.168.2.41 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.1.0.2708 Proxy-Authorization: Digest username="2400", realm="asterisk", nonce="7445a374", uri="sip:9XXXXXXXXXX@192.168.2.150:5060;user=phone", response="6459a5119f1001fa41aefca3bcdc40e7", algorithm=MD5 Max-Forwards: 70 Content-Length: 0 <-------------> [Apr 11 10:39:25] --- (12 headers 0 lines) --- [Apr 11 10:39:26] WARNING[21300]: app_voicemail.c:2299 inboxcount: Failed to obtain database object for ''! [Apr 11 10:39:28] <--- SIP read from 192.168.2.41:5060 ---> BYE sip:9XXXXXXXXXX@192.168.2.150 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.41;branch=z9hG4bKe3eea8286EFCF773 From: "2400" ;tag=8FFD36C6-24E79537 To: ;tag=as0fd29470 CSeq: 3 BYE Call-ID: 1132e582-c34ae83c-7628e665@192.168.2.41 Contact: User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.1.0.2708 Proxy-Authorization: Digest username="2400", realm="asterisk", nonce="7445a374", uri="sip:9XXXXXXXXXX@192.168.2.150:5060;user=phone", response="ee7bca0b95e1f0536d4951e4e3c002b1", algorithm=MD5 Max-Forwards: 70 Content-Length: 0 <-------------> [Apr 11 10:39:28] --- (11 headers 0 lines) --- [Apr 11 10:39:28] Sending to 192.168.2.41 : 5060 (no NAT) [Apr 11 10:39:28] <--- Transmitting (no NAT) to 192.168.2.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.41;branch=z9hG4bKe3eea8286EFCF773;received=192.168.2.41 From: "2400" ;tag=8FFD36C6-24E79537 To: ;tag=as0fd29470 Call-ID: 1132e582-c34ae83c-7628e665@192.168.2.41 CSeq: 3 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Apr 11 10:39:28] -- Hungup 'Zap/1-1' [Apr 11 10:39:28] == Spawn extension (sip-internal, 9XXXXXXXXXX, 2) exited non-zero on 'SIP/2400-08228e58' [Apr 11 10:39:29] Really destroying SIP dialog '1132e582-c34ae83c-7628e665@192.168.2.41' Method: BYE [Apr 11 10:39:37] WARNING[21300]: app_voicemail.c:2299 inboxcount: Failed to obtain database object for ''! [Apr 11 10:39:48] WARNING[21300]: app_voicemail.c:2299 inboxcount: Failed to obtain database object for ''! [Apr 11 10:39:59] WARNING[21300]: app_voicemail.c:2299 inboxcount: Failed to obtain database object for ''! *CLI> exit No such command 'exit' (type 'help' for help) *CLI> exit No such command 'exit' (type 'help' for help) *CLI> quit No such command 'quit' (type 'help' for help) *CLI> stio[Apr 11 10:40:10] WARNING[21300]: app_voicemail.c:2299 inboxcount: Failed to obtain database object for ''!  *CLI> stop now [Apr 11 10:40:14] Beginning asterisk shutdown.... [Apr 11 10:40:14] Executing last minute cleanups [Apr 11 10:40:14] == Destroying musiconhold processes [Apr 11 10:40:14] Asterisk cleanly ending (0).