ASTERISK SERVER = AAA.AAA.AAA.AAA OUT GOING ROUTE = BBB.BBB.BBB.BBB CISCO ATA IP = CCC.CCC.CCC.CCC GATEWAY IP = DDD.DDD.DDD.DDD Date: Sun, 11 Feb 2007 16:21:37 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- Destroying call '01aa801751b812ad2e5bc98331e6fcc6@AAA.AAA.AAA.AAA' -- SIP Seeding peer from astdb: '101038' at 101038@DDD.DDD.DDD.DDD:51062 for 3600 12 headers, 0 lines Reliably Transmitting (NAT) to DDD.DDD.DDD.DDD:51062: OPTIONS sip:101038@CCC.CCC.CCC.CCC:8891;user=phone;transport=udp SIP/2.0 Via: SIP/2.0/UDP AAA.AAA.AAA.AAA:5060;branch=z9hG4bK6468304d;rport From: "asterisk" ;tag=as1a6cb282 To: Contact: Call-ID: 250c73cb7eaa91ac017a000d65609fd2@AAA.AAA.AAA.AAA CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Sun, 11 Feb 2007 16:21:37 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- Destroying call '250c73cb7eaa91ac017a000d65609fd2@AAA.AAA.AAA.AAA' We're at AAA.AAA.AAA.AAA port 15652 Adding codec 0x4 (ulaw) to SDP Adding codec 0x400 (ilbc) to SDP Adding codec 0x1 (g723) to SDP Adding codec 0x100 (g729) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x10 (g726) to SDP Adding codec 0x20 (adpcm) to SDP Adding codec 0x40 (slin) to SDP Adding codec 0x80 (lpc10) to SDP Adding codec 0x200 (speex) to SDP Adding non-codec 0x1 (telephone-event) to SDP 13 headers, 21 lines Reliably Transmitting (NAT) to DDD.DDD.DDD.DDD:51062: INVITE sip:101038@CCC.CCC.CCC.CCC:8891;user=phone;transport=udp SIP/2.0 Via: SIP/2.0/UDP AAA.AAA.AAA.AAA:5060;branch=z9hG4bK17fab459;rport From: "8009598923" ;tag=as469ca43d To: Contact: Call-ID: 41f0016844c24bdf40c5c2c26e852f64@AAA.AAA.AAA.AAA CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Sun, 11 Feb 2007 16:21:37 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 490 v=0 o=root 13205 13205 IN IP4 AAA.AAA.AAA.AAA s=session c=IN IP4 AAA.AAA.AAA.AAA t=0 0 m=audio 15652 RTP/AVP 0 97 4 18 3 8 111 5 10 7 110 101 a=rtpmap:0 PCMU/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:7 LPC/8000 a=rtpmap:110 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Called 101038 Retransmitting #1 (NAT) to DDD.DDD.DDD.DDD:51062: INVITE sip:101038@CCC.CCC.CCC.CCC:8891;user=phone;transport=udp SIP/2.0 Via: SIP/2.0/UDP AAA.AAA.AAA.AAA:5060;branch=z9hG4bK17fab459;rport From: "8009598923" ;tag=as469ca43d To: Contact: Call-ID: 41f0016844c24bdf40c5c2c26e852f64@AAA.AAA.AAA.AAA CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Sun, 11 Feb 2007 16:21:37 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 490 v=0 o=root 13205 13205 IN IP4 AAA.AAA.AAA.AAA s=session c=IN IP4 AAA.AAA.AAA.AAA t=0 0 m=audio 15652 RTP/AVP 0 97 4 18 3 8 111 5 10 7 110 101 a=rtpmap:0 PCMU/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:7 LPC/8000 a=rtpmap:110 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Feb 11 10:21:39 NOTICE[13437]: chan_sip.c:11200 handle_request_register: Registration from '143000 ' failed for '70.154.21.166' - Username/auth name mismatch Retransmitting #2 (NAT) to DDD.DDD.DDD.DDD:51062: INVITE sip:101038@CCC.CCC.CCC.CCC:8891;user=phone;transport=udp SIP/2.0 Via: SIP/2.0/UDP AAA.AAA.AAA.AAA:5060;branch=z9hG4bK17fab459;rport From: "8009598923" ;tag=as469ca43d To: Contact: Call-ID: 41f0016844c24bdf40c5c2c26e852f64@AAA.AAA.AAA.AAA CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Sun, 11 Feb 2007 16:21:37 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 490 v=0 o=root 13205 13205 IN IP4 AAA.AAA.AAA.AAA s=session c=IN IP4 AAA.AAA.AAA.AAA t=0 0 m=audio 15652 RTP/AVP 0 97 4 18 3 8 111 5 10 7 110 101 a=rtpmap:0 PCMU/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:7 LPC/8000 a=rtpmap:110 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Retransmitting #3 (NAT) to DDD.DDD.DDD.DDD:51062: INVITE sip:101038@CCC.CCC.CCC.CCC:8891;user=phone;transport=udp SIP/2.0 Via: SIP/2.0/UDP AAA.AAA.AAA.AAA:5060;branch=z9hG4bK17fab459;rport From: "8009598923" ;tag=as469ca43d To: Contact: Call-ID: 41f0016844c24bdf40c5c2c26e852f64@AAA.AAA.AAA.AAA CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Sun, 11 Feb 2007 16:21:37 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 490 v=0 o=root 13205 13205 IN IP4 AAA.AAA.AAA.AAA s=session c=IN IP4 AAA.AAA.AAA.AAA t=0 0 m=audio 15652 RTP/AVP 0 97 4 18 3 8 111 5 10 7 110 101 a=rtpmap:0 PCMU/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:7 LPC/8000 a=rtpmap:110 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Feb 11 10:21:41 NOTICE[13437]: chan_sip.c:2010 auto_congest: Auto-congesting SIP/101038-0889b200 -- SIP/101038-0889b200 is circuit-busy Scheduling destruction of call '41f0016844c24bdf40c5c2c26e852f64@AAA.AAA.AAA.AAA' in 32000 ms -- SIP Seeding peer from astdb: '101038' at 101038@DDD.DDD.DDD.DDD:51062 for 3600 12 headers, 0 lines Reliably Transmitting (NAT) to DDD.DDD.DDD.DDD:51062: OPTIONS sip:101038@CCC.CCC.CCC.CCC:8891;user=phone;transport=udp SIP/2.0 Via: SIP/2.0/UDP AAA.AAA.AAA.AAA:5060;branch=z9hG4bK648764bc;rport From: "asterisk" ;tag=as5dd3b2a9 To: Contact: Call-ID: 7b327fc71584729f14151cbf4cc1f6ff@AAA.AAA.AAA.AAA CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Sun, 11 Feb 2007 16:21:41 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- Destroying call '7b327fc71584729f14151cbf4cc1f6ff@AAA.AAA.AAA.AAA' == Everyone is busy/congested at this time (1:0/1/0) -- Executing VoiceMail("SIP/101033-08895cc0", "b101038") in new stack -- SIP Seeding peer from astdb: '101033' at 101033@DDD.DDD.DDD.DDD:51062 for 3600 12 headers, 0 lines Reliably Transmitting (NAT) to DDD.DDD.DDD.DDD:51062: OPTIONS sip:101033@CCC.CCC.CCC.CCC:8891;user=phone;transport=udp SIP/2.0 Via: SIP/2.0/UDP AAA.AAA.AAA.AAA:5060;branch=z9hG4bK02c3e4fb;rport From: "asterisk" ;tag=as7f5abd72 To: Contact: Call-ID: 554f67c4164758093870e0e0141d87e7@AAA.AAA.AAA.AAA CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Sun, 11 Feb 2007 16:21:41 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- Destroying call '554f67c4164758093870e0e0141d87e7@AAA.AAA.AAA.AAA' Feb 11 10:21:41 WARNING[13914]: app_voicemail.c:2461 leave_voicemail: No entry in voicemail config file for '101038' -- Executing Hangup("SIP/101033-08895cc0", "") in new stack == Spawn extension (PBX, 101038, 103) exited non-zero on 'SIP/101033-08895cc0' -- SIP Seeding peer from astdb: '101033' at 101033@DDD.DDD.DDD.DDD:51062 for 3600 12 headers, 0 lines Reliably Transmitting (NAT) to DDD.DDD.DDD.DDD:51062: OPTIONS sip:101033@CCC.CCC.CCC.CCC:8891;user=phone;transport=udp SIP/2.0 Via: SIP/2.0/UDP AAA.AAA.AAA.AAA:5060;branch=z9hG4bK60435e42;rport From: "asterisk" ;tag=as3cc9888c To: Contact: Call-ID: 116b197060933f9f512bfc57233a639c@AAA.AAA.AAA.AAA CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Sun, 11 Feb 2007 16:21:41 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- Destroying call '116b197060933f9f512bfc57233a639c@AAA.AAA.AAA.AAA' [PBX] exten => *100,1,DeadAGI,SPSayBalance.pl|${EXTEN}|${ACCOUNTCODE} exten => 100,1,DeadAGI,SPSayBalance.pl|${EXTEN}|${ACCOUNTCODE} exten => _1XXXXX,1,Dial(SIP/${EXTEN},1000,tr) exten => _1XXXXX,2,Voicemail(b${EXTEN}) exten => _1XXXXX,3,Hangup exten => _1XXXXX,102,Voicemail(b${EXTEN}) exten => _1XXXXX,103,Hangup exten => _2XXXXX,1,Dial(SIP/${EXTEN},1000,tr) exten => _2XXXXX,2,Voicemail(b${EXTEN}) exten => _2XXXXX,3,Hangup exten => _2XXXXX,102,Voicemail(b${EXTEN}) exten => _2XXXXX,103,Hangup exten => _300XXXXXXX,1,Dial(SIP/${EXTEN}@phoneopia.com) exten => _400XXXXXXX,1,Dial(SIP/${EXTEN}@phoneopia.com) exten => 111,1,Dial(SIP/101010@ip-pabx.com:8891) exten => *123,1,Answer exten => *123,2,Wait(2) exten => *123,3,VoicemailMain(${CALLERIDNUM}) exten => *100,1,DeadAGI(SayBalance.pl,${CALLERIDNUM},${EXTEN}) exten => _78XXXX,1,Dial(SIP/${EXTEN}@superphone.us,100,tr) exten => _78XXXX,2,Hangup exten => _68XXXX,1,Dial(SIP/${EXTEN}@superphone.us,100,tr) exten => _68XXXX,2,Hangup ;exten => _X.,1,DeadAGI(pbxOut.pl,${CALLERIDNUM},${EXTEN}) ;exten => _X.,2,Hangup exten => _X.,1,DeadAGI(pbxOut.pl,${ACCOUNTCODE},${EXTEN}) exten => _X.,2,Hangup