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. ========================================================================= == Parsing '/etc/asterisk/asterisk.conf': == Found == Parsing '/etc/asterisk/extconfig.conf': == Found == Binding queue_log to mysql/asterisk/queue_log Connected to Asterisk 1.6.2.1 currently running on asterisk1621 (pid = 719) asterisk1621*CLI> Verbosity is at least 6 asterisk1621*CLI> core cshow ch channel channels channeltypes channeltype asterisk1621*CLI> core show channels asterisk1621*CLI> Channel Location State Application(Data) 0 active channels 0 active calls 24 calls processed asterisk1621*CLI> core show channelsexitsip set debug ip 192.168.0.3 asterisk1621*CLI> SIP Debugging Enabled for IP: 192.168.0.3 asterisk1621*CLI> sip set debug ip 192.168.0.3core show channelsexitsip set debug ip 192.168.0.3exitsip set debug ip 192.168.0.3peer 13456789 asterisk1621*CLI> SIP Debugging Enabled for IP: 192.168.0.3:5060 asterisk1621*CLI> [100201-150128] WARNING[742]: netsock.c:159 ast_netsock_set_qos: Unable to set SIP RTP TOS to 184, may be you have no root privileges == Using SIP RTP CoS mark 5 == Using SIP VRTP CoS mark 6 [100201-150128] WARNING[742]: netsock.c:159 ast_netsock_set_qos: Unable to set UDPTL TOS to 184, may be you have no root privileges == Using UDPTL CoS mark 5 Sending to 192.168.0.3 : 5060 (no NAT) Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 18 Found RTP audio format 99 Found RTP audio format 97 Found RTP audio format 19 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G729 for ID 18 Found audio description format G726-32 for ID 99 Found audio description format telephone-event for ID 97 Found audio description format CN for ID 19 Capabilities: us - 0x3c030e (gsm|ulaw|alaw|g729|speex|h261|h263|h263p|h264), peer - audio=0x90c (ulaw|alaw|g726|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x10c (ulaw|alaw|g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x3 (telephone-event|CN), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.0.3:16468 Peer doesn't provide video Looking for 3456789 in in-13456789 (domain 192.168.150.2) list_route: hop: <--- Transmitting (no NAT) to 192.168.0.3:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.3:5060;branch=z9hG4bK25219A3;received=192.168.0.3 From: ;tag=42E0B68-8D6 To: Call-ID: 7FFB32D4-EA311DF-830DA116-E5E7AB98@192.168.0.3 CSeq: 101 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 <------------> -- Executing [3456789@in-13456789:1] Set("SIP/13456789-0000002d", "CDR(intrunk)=13456789") in new stack -- Executing [3456789@in-13456789:2] Goto("SIP/13456789-0000002d", "welcome_msg,s,1") in new stack -- Goto (welcome_msg,s,1) -- Executing [s@welcome_msg:1] Ringing("SIP/13456789-0000002d", "") in new stack <--- Transmitting (no NAT) to 192.168.0.3:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.0.3:5060;branch=z9hG4bK25219A3;received=192.168.0.3 From: ;tag=42E0B68-8D6 To: ;tag=as77dfffb4 Call-ID: 7FFB32D4-EA311DF-830DA116-E5E7AB98@192.168.0.3 CSeq: 101 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 <------------> -- Executing [s@welcome_msg:2] Set("SIP/13456789-0000002d", "TIMEOUT(digit)=3") in new stack -- Digit timeout set to 3.000 -- Executing [s@welcome_msg:3] SetMusicOnHold("SIP/13456789-0000002d", "default") in new stack -- Executing [s@welcome_msg:4] Set("SIP/13456789-0000002d", "DIAL_OPTIONS=m") in new stack -- Executing [s@welcome_msg:5] Set("SIP/13456789-0000002d", "TIMEOUT(absolute)=3600") in new stack Channel will hangup at 2010-02-01 16:01:28.211 COT. -- Executing [s@welcome_msg:6] BackGround("SIP/13456789-0000002d", "pbx1") in new stack Audio is at 192.168.150.2 port 13222 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 192.168.0.3:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.3:5060;branch=z9hG4bK25219A3;received=192.168.0.3 From: ;tag=42E0B68-8D6 To: ;tag=as77dfffb4 Call-ID: 7FFB32D4-EA311DF-830DA116-E5E7AB98@192.168.0.3 CSeq: 101 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-Type: application/sdp Content-Length: 333 v=0 o=root 1625233331 1625233331 IN IP4 192.168.150.2 s=Asterisk PBX 1.6.2.1 c=IN IP4 192.168.150.2 t=0 0 m=audio 13222 RTP/AVP 0 8 18 97 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> asterisk1621*CLI> -- Playing 'pbx1.slin' (language 'es') asterisk1621*CLI> == CDR updated on SIP/13456789-0000002d -- Executing [9800@welcome_msg:1] Answer("SIP/13456789-0000002d", "") in new stack -- Executing [9800@welcome_msg:2] Wait("SIP/13456789-0000002d", "1") in new stack asterisk1621*CLI> -- Executing [9800@welcome_msg:3] MusicOnHold("SIP/13456789-0000002d", "default") in new stack -- Started music on hold, class 'default', on channel 'SIP/13456789-0000002d' asterisk1621*CLI> [100201-150221] NOTICE[746]: res_musiconhold.c:603 monmp3thread: Request to schedule in the past?!?! [100201-150221] WARNING[746]: res_musiconhold.c:436 spawn_mp3: /usr/share/asterisk/mohprueba is not a valid directory [100201-150221] WARNING[746]: res_musiconhold.c:578 monmp3thread: Unable to spawn mp3player asterisk1621*CLI> asterisk1621*CLI> [100201-150309] WARNING[742]: netsock.c:159 ast_netsock_set_qos: Unable to set SIP RTP TOS to 184, may be you have no root privileges == Using SIP RTP CoS mark 5 == Using SIP VRTP CoS mark 6 [100201-150309] WARNING[742]: netsock.c:159 ast_netsock_set_qos: Unable to set UDPTL TOS to 184, may be you have no root privileges == Using UDPTL CoS mark 5 asterisk1621*CLI> [100201-150319] WARNING[742]: chan_sip.c:3779 retrans_pkt: Maximum retries exceeded on transmission 6033ed7368a698b8@172.30.0.44 for seqno 37155 (Critical Response) -- See doc/sip-retransmit.txt. asterisk1621*CLI> set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.0.3, port 5060 Audio is at 192.168.150.2 port 13222 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.0.3:5060: INVITE sip:18765432@192.168.0.3:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.150.2:5060;branch=z9hG4bK2377a139;rport Max-Forwards: 70 From: ;tag=as77dfffb4 To: ;tag=42E0B68-8D6 Contact: Call-ID: 7FFB32D4-EA311DF-830DA116-E5E7AB98@192.168.0.3 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.2.1 Require: timer Session-Expires: 1800;refresher=uas Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer X-asterisk-Info: SIP re-invite (Session-Timers) Content-Type: application/sdp Content-Length: 333 v=0 o=root 1625233331 1625233331 IN IP4 192.168.150.2 s=Asterisk PBX 1.6.2.1 c=IN IP4 192.168.150.2 t=0 0 m=audio 13222 RTP/AVP 0 8 18 97 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- asterisk1621*CLI> <--- SIP read from UDP:192.168.0.3:5060 ---> SIP/2.0 422 Session Timer too small Via: SIP/2.0/UDP 192.168.150.2:5060;branch=z9hG4bK2377a139;rport From: ;tag=as77dfffb4 To: ;tag=42E0B68-8D6 Call-ID: 7FFB32D4-EA311DF-830DA116-E5E7AB98@192.168.0.3 Min-SE: 3600 CSeq: 102 INVITE Content-Length: 0 <-------------> --- (8 headers 0 lines) --- set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.0.3, port 5060 Transmitting (no NAT) to 192.168.0.3:5060: ACK sip:18765432@192.168.0.3:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.150.2:5060;branch=z9hG4bK2377a139;rport Max-Forwards: 70 From: ;tag=as77dfffb4 To: ;tag=42E0B68-8D6 Contact: Call-ID: 7FFB32D4-EA311DF-830DA116-E5E7AB98@192.168.0.3 CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.2.1 Content-Length: 0 --- Audio is at 192.168.150.2 port 13222 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.0.3:5060: INVITE sip:13456789@ SIP/2.0 Via: SIP/2.0/UDP 192.168.150.2:5060;branch=z9hG4bK76920569;rport Max-Forwards: 70 Route: From: "18765432" ;tag=as77dfffb4 To: Contact: Call-ID: 7FFB32D4-EA311DF-830DA116-E5E7AB98@192.168.0.3 CSeq: 103 INVITE User-Agent: Asterisk PBX 1.6.2.1 Date: Mon, 01 Feb 2010 20:16:28 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 333 v=0 o=root 1625233331 1625233332 IN IP4 192.168.150.2 s=Asterisk PBX 1.6.2.1 c=IN IP4 192.168.150.2 t=0 0 m=audio 13222 RTP/AVP 0 8 18 97 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- asterisk1621*CLI> <--- SIP read from UDP:192.168.0.3:5060 ---> SIP/2.0 400 Bad Request - 'Malformed/Missing URL' Via: SIP/2.0/UDP 192.168.150.2:5060;branch=z9hG4bK76920569;rport From: "18765432" ;tag=as77dfffb4 To: ;tag=43BC730-2703;tag=43BC730-2703 Date: Mon, 01 Feb 2010 20:17:03 GMT Call-ID: 7FFB32D4-EA311DF-830DA116-E5E7AB98@192.168.0.3 Server: Cisco-SIPGateway/IOS-12.x CSeq: 103 INVITE Allow-Events: telephone-event Reason: Q.850;cause=100 Content-Length: 0 <-------------> --- (11 headers 0 lines) --- -- Got SIP response 400 "Bad Request - 'Malformed/Missing URL'" back from 192.168.0.3 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.0.3, port 5060 Transmitting (no NAT) to 192.168.0.3:5060: ACK sip:18765432@192.168.0.3:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.150.2:5060;branch=z9hG4bK76920569;rport Max-Forwards: 70 From: ;tag=as77dfffb4 To: ;tag=42E0B68-8D6 Contact: Call-ID: 7FFB32D4-EA311DF-830DA116-E5E7AB98@192.168.0.3 CSeq: 103 ACK User-Agent: Asterisk PBX 1.6.2.1 Content-Length: 0 --- asterisk1621*CLI> <--- SIP read from UDP:192.168.0.3:5060 ---> SIP/2.0 400 Bad Request - 'Malformed/Missing URL' Via: SIP/2.0/UDP 192.168.150.2:5060;branch=z9hG4bK76920569;rport From: "18765432" ;tag=as77dfffb4 To: ;tag=43BC730-2703;tag=43BC730-2703 Call-ID: 7FFB32D4-EA311DF-830DA116-E5E7AB98@192.168.0.3 CSeq: 103 INVITE Reason: Q.850;cause=100 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- asterisk1621*CLI> <--- SIP read from UDP:192.168.0.3:5060 ---> SIP/2.0 400 Bad Request - 'Malformed/Missing URL' Via: SIP/2.0/UDP 192.168.150.2:5060;branch=z9hG4bK76920569;rport From: "18765432" ;tag=as77dfffb4 To: ;tag=43BC730-2703;tag=43BC730-2703;tag=43BC730-2703 Call-ID: 7FFB32D4-EA311DF-830DA116-E5E7AB98@192.168.0.3 CSeq: 103 INVITE Reason: Q.850;cause=100 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- asterisk1621*CLI> <--- SIP read from UDP:192.168.0.3:5060 ---> SIP/2.0 400 Bad Request - 'Malformed/Missing URL' Via: SIP/2.0/UDP 192.168.150.2:5060;branch=z9hG4bK76920569;rport From: "18765432" ;tag=as77dfffb4 To: ;tag=43BC730-2703;tag=43BC730-2703;tag=43BC730-2703;tag=43BC730-2703 Call-ID: 7FFB32D4-EA311DF-830DA116-E5E7AB98@192.168.0.3 CSeq: 103 INVITE Reason: Q.850;cause=100 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- asterisk1621*CLI> <--- SIP read from UDP:192.168.0.3:5060 ---> SIP/2.0 400 Bad Request - 'Malformed/Missing URL' Via: SIP/2.0/UDP 192.168.150.2:5060;branch=z9hG4bK76920569;rport From: "18765432" ;tag=as77dfffb4 To: ;tag=43BC730-2703;tag=43BC730-2703;tag=43BC730-2703;tag=43BC730-2703;tag=43BC730-2703 Call-ID: 7FFB32D4-EA311DF-830DA116-E5E7AB98@192.168.0.3 CSeq: 103 INVITE Reason: Q.850;cause=100 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- asterisk1621*CLI> <--- SIP read from UDP:192.168.0.3:5060 ---> SIP/2.0 400 Bad Request - 'Malformed/Missing URL' Via: SIP/2.0/UDP 192.168.150.2:5060;branch=z9hG4bK76920569;rport From: "18765432" ;tag=as77dfffb4 To: ;tag=43BC730-2703;tag=43BC730-2703;tag=43BC730-2703;tag=43BC730-2703;tag=43BC730-2703;tag=43BC730-2703 Call-ID: 7FFB32D4-EA311DF-830DA116-E5E7AB98@192.168.0.3 CSeq: 103 INVITE Reason: Q.850;cause=100 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- asterisk1621*CLI> <--- SIP read from UDP:192.168.0.3:5060 ---> SIP/2.0 400 Bad Request - 'Malformed/Missing URL' Via: SIP/2.0/UDP 192.168.150.2:5060;branch=z9hG4bK76920569;rport From: "18765432" ;tag=as77dfffb4 To: ;tag=43BC730-2703;tag=43BC730-2703;tag=43BC730-2703;tag=43BC730-2703;tag=43BC730-2703;tag=43BC730-2703;tag=43BC730-2703 Call-ID: 7FFB32D4-EA311DF-830DA116-E5E7AB98@192.168.0.3 CSeq: 103 INVITE Reason: Q.850;cause=100 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- asterisk1621*CLI> exit Sending to 192.168.0.3 : 5060 (no NAT) <--- Transmitting (no NAT) to 192.168.0.3:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.3:5060;branch=z9hG4bK25E16E9;received=192.168.0.3 From: ;tag=42E0B68-8D6 To: ;tag=as77dfffb4 Call-ID: 7FFB32D4-EA311DF-830DA116-E5E7AB98@192.168.0.3 CSeq: 102 BYE Server: Asterisk PBX 1.6.2.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 asterisk1621*CLI> exit <------------> -- Stopped music on hold on SIP/13456789-0000002d == Spawn extension (welcome_msg, 9800, 3) exited non-zero on 'SIP/13456789-0000002d' asterisk1621*CLI> exit Executing last minute cleanups