Script started on Sat 03 Jun 2006 02:48:15 PM PDT [root@localhost tftpboot]# exec asterisk -r Asterisk 1.2.7.1, Copyright (C) 1999 - 2006 Digium, Inc. and others. Created by Mark Spencer Asterisk comes with ABSOLUTELY NO WARRANTY; type '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 'show license' for details. ========================================================================= Connected to Asterisk 1.2.7.1 currently running on localhost (pid = 2076) localhost*CLI> Verbosity is at least 11 Core debug is at least 5 localhost*CLI> <-- SIP read from 192.168.233.237:5060: INVITE sip:43235899999@192.168.233.235:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.233.237;branch=z9hG4bKfa5534ac7 Max-Forwards: 70 Content-Length: 590 To: 43235899999 From: 102 ;tag=931535749fef423 Call-ID: c4fafd9f5cbb535473bf4db88cd00c43@192.168.233.237 CSeq: 1148010823 INVITE Supported: timer Allow-Events: talk,hold,conference Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO Content-Type: application/sdp Contact: 102 Supported: replaces User-Agent: Aastra 480i Cordless/1.4.0.1048 Brcm Callctrl/1.5 MxSF/v3.2.6.26 v=0 o=MxSIP 0 294373501 IN IP4 192.168.233.237 s=SIP Call c=IN IP4 192.168.233.237 t=0 0 m=audio 3000 RTP/AVP 0 18 96 102 107 104 105 106 4 97 98 2 99 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:96 BV16/8000 a=rtpmap:102 BV32/16000 a=rtpmap:107 L16/16000 a=rtpmap:104 PCMU/16000 a=rtpmap:105 PCMA/16000 a=rtpmap:106 L16/8000 a=rtpmap:4 G723/8000 a=rtpmap:97 G726-16/8000 a=rtpmap:98 G726-24/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:99 G726-40/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=silenceSupp:on - - - - --- (15 headers 24 lines)--- Using INVITE request as basis request - c4fafd9f5cbb535473bf4db88cd00c43@192.168.233.237 Sending to 192.168.233.237 : 5060 (non-NAT) Reliably Transmitting (no NAT) to 192.168.233.237:5060: SIP/2.0 407 Proxy Authentication Required V localhost*CLI> ia: SIP/2.0/UDP 192.168.233.237;branch=z9hG4bKfa5534ac7;received=192.168.233.237 From: 102 ;tag=931535749fef423 To: 43235899999 ;tag=as16d77c39 Call-ID: c4fafd9f5cbb535473bf4db88cd00c43@192.168.233.237 CSeq: 1148010823 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Proxy-Authenticate: Digest realm="asterisk", nonce="535a6aef" Content-Length: 0 --- Scheduling destruction of call 'c4fafd9f5cbb535473bf4db88cd00c43@192.168.233.237' in 15000 ms Found user '102' localhost*CLI> <-- SIP read from 192.168.233.237:5060: ACK sip:4323NXXXXXX@192.168.233.235:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.233.237;branch=z9hG4bKfa5534ac7 Max-Forwards: 70 Content-Length: 0 To: 4323NXXXXXX ;tag=as16d77c39 From: 102 ;tag=931535749fef423 Call-ID: c4fafd9f5cbb535473bf4db88cd00c43@192.168.233.237 CSeq: 1148010823 ACK User-Agent: Aastra 480i Cordless/1.4.0.1048 Brcm Callctrl/1.5 MxSF/v3.2.6.26 --- (9 headers 0 lines)--- localhost*CLI> <-- SIP read from 192.168.233.237:5060: INVITE sip:4323NXXXXXX@192.168.233.235:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.233.237;branch=z9hG4bK618118c77 Max-Forwards: 70 Content-Length: 590 To: 4323NXXXXXX From: 102 ;tag=931535749fef423 Call-ID: c4fafd9f5cbb535473bf4db88cd00c43@192.168.233.237 CSeq: 1148010824 INVITE Supported: timer Allow-Events: talk,hold,conference Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO Contact: 102 Content-Type: application/sdp Supported: replaces Proxy-Authorization:Digest response="b3117edb28e777fb5717b4640c872e66",username="102",realm="asterisk",nonce="535a6aef",uri="sip:4323NXXXXXX@192.168.233.235:5060" User-Agent: Aastra 480i Cordless/1.4.0.1048 Brcm Callctrl/1.5 MxSF/v3.2.6.26 v=0 o=MxSIP 0 294373501 IN IP4 192.168.233.237 s=SIP Call c=IN IP4 192.168.233.237 t=0 0 m=audio 3000 RTP/AVP 0 18 96 102 107 104 105 106 4 97 98 2 99 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:96 BV16/8000 a=rtpmap:102 BV32/16000 a=rtpmap:107 L16/16000 a=rtpmap:104 PCMU/16000 a=rtpmap:105 PCMA/16000 a=rtpmap:106 L16/8000 a=rtpmap:4 G723/8000 a=rtpmap:97 G726-16/8000 a=rtpmap:98 G726-24/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:99 G726-40/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=silenceSupp:on - - - - --- (16 headers 24 lines)--- Using INVITE request as basis request - c4fafd9f5cbb535473bf4db88cd00c43@192.168.233.237 Sending to 192.168.233.237 : 5060 (non-NAT) Found user '102' Found RTP audio format 0 Found RTP audio format 18 Found RTP audio format 96 Found RTP audio format 102 Found RTP audio format 107 Found RTP audio format 104 Found RTP audio format 105 Found RTP audio format 106 Found RTP audio format 4 Found RTP audio format 97 Found RTP audio format 98 Found RTP audio format 2 Found RTP audio format 99 Found RTP audio format 8 Found RTP audio format 101 localhost*CLI> Peer audio RTP is at port 192.168.233.237:3000 Found description format PCMU Found description format G729 Found description format BV16 Found description format BV32 Found description format L16 Found description format PCMU Found description format PCMA Found description format L16 Found description format G723 Found description format G726-16 Found description format G726-24 Found description format G726-32 Found description format G726-40 Found description format PCMA Found description format telephone-event Capabilities: us - 0x4 (ulaw), peer - audio=0x55d (g723|ulaw|alaw|g726|slin|g729|ilbc)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 4323NXXXXXX in internal111 (domain 192.168.233.235) list_route: hop: Transmitting (no NAT) to 192.168.233.237:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.233.237;branch=z9hG4bK618118c77;received=192.168.233.237 From: 102 ;tag=931535749fef423 To: 4323NXXXXXX Call-ID: c4fafd9f5cbb535473bf4db88cd00c43@192.168.233.237 CSeq: 1148010824 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- localhost*CLI> Reliably Transmitting (no NAT) to 192.168.233.240:5060: NOTIFY sip:103@192.168.233.240 SIP/2.0 Via: SIP/2.0/UDP 192.168.233.235:5060;branch=z9hG4bK301315de From: Ira ;tag=as14e35da4 To: 103 ;tag=535790d98069983 Contact: Call-ID: 27f7d962409c15d2ac351ca5987858ac@192.168.233.240 CSeq: 105 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 210 confirmed --- Extension Changed 102 new state InUse for Notify User 103 Reliably Transmitting (no NAT) to 192.168.233.233:5060: NOTIFY sip:101@192.168.233.233 SIP/2.0 Via: SIP/2.0/UDP 192.168.233.235:5060;branch=z9hG4bK21f14b18 From: Ira ;tag=as6e4a139b To: 101 ;tag=57883542f06cd06 Contact: Call-ID: 5b30f7b60c1b618505c2d4b30159808b@192.168.233.233 CSeq: 107 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 210 confirmed --- Extension Changed 102 new state InUse for Notify User 101 localhost*CLI> -- Executing Macro("SIP/102-854a", "dial-fish|4|1323NXXXXXX") in new stack localhost*CLI> -- Executing NoOp("SIP/102-854a", "macro-dial-fish") in new stack localhost*CLI> -- Executing Set("SIP/102-854a", "CALLERID(Name)=WorldPeace") in new stack localhost*CLI> -- Executing Set("SIP/102-854a", "CALLERID(Number)=3104990989") in new stack localhost*CLI> -- Executing Dial("SIP/102-854a", "SIP/fishtune/1323NXXXXXX||j") in new stack localhost*CLI> We're at 24.aaa.bbb.7 port 16000 localhost*CLI> Adding codec 0x4 (ulaw) to SDP localhost*CLI> Adding non-codec 0x1 (telephone-event) to SDP localhost*CLI> 13 headers, 10 lines localhost*CLI> Reliably Transmitting (NAT) to 4.79.19.59:5060: INVITE sip:1323NXXXXXX@gw4.telasip.com SIP/2.0 Via: SIP/2.0/UDP 24.aaa.bbb.7:5060;branch=z9hG4bK71ac24f1;rport From: "WorldPeace" ;tag=as7a631e92 To: Contact: Call-ID: 252cfa1e2237cf1849fe66a66bcc2d7b@24.aaa.bbb.7 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Sat, 03 Jun 2006 21:48:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 212 v=0 o=root 2076 2076 IN IP4 24.aaa.bbb.7 s=session c=IN IP4 24.aaa.bbb.7 t=0 0 m=audio 16000 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- localhost*CLI> -- Called fishtune/1323NXXXXXX localhost*CLI> <-- SIP read from 192.168.233.233:5060: SIP/2.0 200 OK Call-ID: 5b30f7b60c1b618505c2d4b30159808b@192.168.233.233 CSeq: 107 NOTIFY From: Ira ;tag=as6e4a139b To: 101 ;tag=57883542f06cd06 Via: SIP/2.0/UDP 192.168.233.235:5060;branch=z9hG4bK21f14b18 Content-Length: 0 Contact: 101 Supported: replaces User-Agent: Aastra 480i Cordless/1.4.0.1048 Brcm Callctrl/1.5 MxSF/v3.2.6.26 localhost*CLI> --- (10 headers 0 lines)--- localhost*CLI> <-- SIP read from 192.168.233.240:5060: SIP/2.0 200 OK Call-ID: 27f7d962409c15d2ac351ca5987858ac@192.168.233.240 CSeq: 105 NOTIFY From: Ira ;tag=as14e35da4 To: 103 ;tag=535790d98069983 Via: SIP/2.0/UDP 192.168.233.235:5060;branch=z9hG4bK301315de Content-Length: 0 Contact: 103 Supported: replaces User-Agent: Aastra 480i Cordless/1.4.0.1048 Brcm Callctrl/1.5 MxSF/v3.2.6.26 localhost*CLI> --- (10 headers 0 lines)--- localhost*CLI> <-- SIP read from 4.79.19.59:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 24.aaa.bbb.7:5060;branch=z9hG4bK71ac24f1;received=24.aaa.bbb.7;rport=5060 From: "WorldPeace" ;tag=as7a631e92 To: ;tag=as1ec10eaf Call-ID: 252cfa1e2237cf1849fe66a66bcc2d7b@24.aaa.bbb.7 CSeq: 102 INVITE User-Agent: Telasip Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Proxy-Authenticate: Digest realm="telasip.com", nonce="5ebb03ea" Content-Length: 0 localhost*CLI> --- (11 headers 0 lines)--- Transmitting (NAT) to 4.79.19.59:5060: ACK sip:1323NXXXXXX@gw4.telasip.com SIP/2.0 Via: SIP/2.0/UDP 24.aaa.bbb.7:5060;branch=z9hG4bK71ac24f1;rport From: "WorldPeace" ;tag=as7a631e92 To: ;tag=as1ec10eaf Contact: Call-ID: 252cfa1e2237cf1849fe66a66bcc2d7b@24.aaa.bbb.7 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- We're at 24.aaa.bbb.7 port 16000 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 4.79.19.59:5060: INVITE sip:1323NXXXXXX@gw4.telasip.com SIP/2.0 Via: SIP/2.0/UDP 24.aaa.bbb.7:5060;branch=z9hG4bK75fdff0c;rport From: "WorldPeace" ;tag=as7a631e92 To: Contact: Call-ID: 252cfa1e2237cf1849fe66a66bcc2d7b@24.aaa.bbb.7 CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Proxy-Authorization: Digest username="xXxXxXxxx", realm="telasip.com", algorithm=MD5, uri="sip:1323NXXXXXX@gw4.telasip.com", nonce="5ebb03ea", response="54ede175f64806ed8b197f02d655d9de", opaque="" Date: Sat, 03 Jun 2006 21:48:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 212 v=0 o=root 2076 2077 IN IP4 24.aaa.bbb.7 s=session c=IN IP4 24.aaa.bbb.7 t=0 0 m=audio 16000 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- localhost*CLI> <-- SIP read from 4.79.19.59:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 24.aaa.bbb.7:5060;branch=z9hG4bK75fdff0c;received=24.aaa.bbb.7;rport=5060 From: "WorldPeace" ;tag=as7a631e92 To: Call-ID: 252cfa1e2237cf1849fe66a66bcc2d7b@24.aaa.bbb.7 CSeq: 103 INVITE User-Agent: Telasip Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- (10 headers 0 lines)--- localhost*CLI> <-- SIP read from 4.79.19.59:5060: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 24.aaa.bbb.7:5060;branch=z9hG4bK75fdff0c;received=24.aaa.bbb.7;rport=5060 From: "WorldPeace" ;tag=as7a631e92 To: ;tag=as30c0b8a1 Call-ID: 252cfa1e2237cf1849fe66a66bcc2d7b@24.aaa.bbb.7 CSeq: 103 INVITE User-Agent: Telasip Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 212 v=0 o=root 24105 24105 IN IP4 4.79.19.59 s=session c=IN IP4 4.79.19.59 t=0 0 m=audio 13768 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- (11 headers 10 lines)--- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 4.79.19.59:13768 Found description format PCMU Found description format telephone-event Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) localhost*CLI> -- SIP/fishtune-b1ba is making progress passing it to SIP/102-854a We're at 192.168.233.235 port 16396 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Transmitting (no NAT) to 192.168.233.237:5060: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.233.237;branch=z9hG4bK618118c77;received=192.168.233.237 From: 102 ;tag=931535749fef423 To: 4323NXXXXXX ;tag=as6f853a09 Call-ID: c4fafd9f5cbb535473bf4db88cd00c43@192.168.233.237 CSeq: 1148010824 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 220 v=0 o=root 2076 2076 IN IP4 192.168.233.235 s=session c=IN IP4 192.168.233.235 t=0 0 m=audio 16396 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- localhost*CLI> Jun 3 14:48:27 NOTICE[2148]: chan_sip.c:5259 sip_reregister: -- Re-registration for xXxXxXxxx@gw4.telasip.com localhost*CLI> REGISTER 13 headers, 0 lines REGISTER attempt 1 to xXxXxXxxx@gw4.telasip.com Reliably Transmitting (no NAT) to 4.79.19.59:5060: REGISTER sip:gw4.telasip.com SIP/2.0 Via: SIP/2.0/UDP 24.aaa.bbb.7:5060;branch=z9hG4bK574104b2;rport From: ;tag=as1c33fd43 To: Call-ID: 126a010c17413f0b2d548857354733ca@127.0.0.1 CSeq: 106 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Authorization: Digest username="xXxXxXxxx", realm="telasip.com", algorithm=MD5, uri="sip:gw4.telasip.com", nonce="7650e1b4", response="32216bd11b0768565929b73d2c1ce1c2", opaque="" Expires: 120 Contact: Event: registration Content-Length: 0 --- localhost*CLI> <-- SIP read from 4.79.19.59:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 24.aaa.bbb.7:5060;branch=z9hG4bK574104b2;received=24.aaa.bbb.7;rport=5060 From: ;tag=as1c33fd43 To: Call-ID: 126a010c17413f0b2d548857354733ca@127.0.0.1 CSeq: 106 REGISTER User-Agent: Telasip Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- (10 headers 0 lines)--- localhost*CLI> <-- SIP read from 4.79.19.59:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 24.aaa.bbb.7:5060;branch=z9hG4bK574104b2;received=24.aaa.bbb.7;rport=5060 From: ;tag=as1c33fd43 To: ;tag=as3595dab6 Call-ID: 126a010c17413f0b2d548857354733ca@127.0.0.1 CSeq: 106 REGISTER User-Agent: Telasip Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: WWW-Authenticate: Digest realm="telasip.com", nonce="28a6ade1" C localhost*CLI> ontent-Length: 0 --- (11 headers 0 lines)--- Responding to challenge, registration to domain/host name gw4.telasip.com REGISTER 13 headers, 0 lines REGISTER attempt 2 to xXxXxXxxx@gw4.telasip.com Reliably Transmitting (no NAT) to 4.79.19.59:5060: REGISTER sip:gw4.telasip.com SIP/2.0 Via: SIP/2.0/UDP 24.aaa.bbb.7:5060;branch=z9hG4bK0cf9a3a5;rport From: ;tag=as1e0ef7be To: Call-ID: 126a010c17413f0b2d548857354733ca@127.0.0.1 CSeq: 107 REGISTER localhost*CLI> User-Agent: Asterisk PBX Max-Forwards: 70 Authorization: Digest username="xXxXxXxxx", realm="telasip.com", algorithm=MD5, uri="sip:gw4.telasip.com", nonce="28a6ade1", response="ec11b2d9c5f41721d863a1c4ecfeb294", opaque="" Expires: 120 Contact: Event: registration Content-Length: 0 --- localhost*CLI> <-- SIP read from 4.79.19.59:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 24.aaa.bbb.7:5060;branch=z9hG4bK0cf9a3a5;received=24.aaa.bbb.7;rport=5060 From: ;tag=as1e0ef7be To: Call-ID: 126a010c17413f0b2d548857354733ca@127.0.0.1 CSeq: 107 REGISTER User-Agent: Telasip Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- (10 headers 0 lines)--- localhost*CLI> <-- SIP read from 4.79.19.59:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 24.aaa.bbb.7:5060;branch=z9hG4bK0cf9a3a5;received=24.aaa.bbb.7;rport=5060 From: ;tag=as1e0ef7be To: ;tag=as3595dab6 Call-ID: 126a010c17413f0b2d548857354733ca@127.0.0.1 CSeq: 107 REGISTER User-Agent: Telasip Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Expires: 120 Contact: ;expires=120 Date: Sat, 03 Jun 2006 21:48:27 GMT Content-Length: 0 v=0 o=root 24105 24106 IN IP4 4.79.19.59 s=session c=IN IP4 4.79.19.59 t=0 0 m=audio 13768 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - localhost*CLI> --- (11 headers 10 lines)--- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 4.79.19.59:13768 Found description format PCMU Found description format telephone-event Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 4.79.19.59, port 5060 Transmitting (NAT) to 4.79.19.59:5060: ACK sip:1323NXXXXXX@4.79.19.59 SIP/2.0 Via: SIP/2.0/UDP 24.aaa.bbb.7:5060;branch=z9hG4bK054ee1f3;rport From: "WorldPeace" ;tag=as7a631e92 To: ;tag=as30c0b8a1 Contact: Call-ID: 252cfa1e2237cf1849fe66a66bcc2d7b@24.aaa.bbb.7 CSeq: 103 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- SIP/fishtune-b1ba answered SIP/102-854a We're at 192.168.233.235 port 16396 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.233.237:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.233.237;branch=z9hG4bK618118c77;received=192.168.233.237 From: 102 ;tag=931535749fef423 To: 4323NXXXXXX ;tag=as6f853a09 Call-ID: c4fafd9f5cbb535473bf4db88cd00c43@192.168.233.237 CSeq: 1148010824 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 220 v=0 o=root 2076 2077 IN IP4 192.168.233.235 s=session c=IN IP4 192.168.233.235 t=0 0 m=audio 16396 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Attempting native bridge of SIP/102-854a and SIP/fishtune-b1ba localhost*CLI> <-- SIP read from 4.79.19.59:5060: BYE sip:3104990989@24.aaa.bbb.7 SIP/2.0 Via: SIP/2.0/UDP 4.79.19.59:5060;branch=z9hG4bK57b7fca8;rport From: ;tag=as30c0b8a1 To: "WorldPeace" ;tag=as7a631e92 Contact: Call-ID: 252cfa1e2237cf1849fe66a66bcc2d7b@24.aaa.bbb.7 CSeq: 102 BYE User-Agent: Telasip Max-Forwards: 70 Content-Length: 0 --- (10 headers 0 lines)--- Sending to 4.79.19.59 : 5060 (NAT) Transmitting (NAT) to 4.79.19.59:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 4.79.19.59:5060;branch=z9hG4bK57b7fca8;received=4.79.19.59;rport=5060 From: ;tag=as30c0b8a1 To: "WorldPeace" ;tag=as7a631e92 Call-ID: 252cfa1e2237cf1849fe66a66bcc2d7b@24.aaa.bbb.7 CSeq: 102 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing localhost*CLI> --- localhost*CLI> Reliably Transmitting (no NAT) to 192.168.233.240:5060: NOTIFY sip:103@192.168.233.240 SIP/2.0 Via: SIP/2.0/UDP 192.168.233.235:5060;branch=z9hG4bK54471662 From: Ira ;tag=as14e35da4 To: 103 ;tag=535790d98069983 Contact: Call-ID: 27f7d962409c15d2ac351ca5987858ac@192.168.233.240 CSeq: 106 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 211 terminated --- Extension Changed 102 new state Idle for Notify User 103 Reliably Transmitting (no NAT) to 192.168.233.233:5060: NOTIFY sip:101@192.168.233.233 SIP/2.0 Via: SIP/2.0/UDP 192.168.233.235:5060;branch=z9hG4bK311ab162 From: Ira ;tag=as6e4a139b To: 101 ;tag=57883542f06cd06 Contact: Call-ID: 5b30f7b60c1b618505c2d4b30159808b@192.168.233.233 CSeq: 108 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 211 terminated --- Extension Changed 102 new state Idle for Notify User 101 localhost*CLI> <-- SIP read from 192.168.233.233:5060: SIP/2.0 200 OK Call-ID: 5b30f7b60c1b618505c2d4b30159808b@192.168.233.233 CSeq: 108 NOTIFY From: Ira ;tag=as6e4a139b To: 101 ;tag=57883542f06cd06 Via: SIP/2.0/UDP 192.168.233.235:5060;branch=z9hG4bK311ab162 Content-Length: 0 Contact: 101 Supported: replaces User-Agent: Aastra 480i Cordless/1.4.0.1048 Brcm Callctrl/1.5 MxSF/v3.2.6.26 --- (10 headers 0 lines)--- Destroying call '252cfa1e2237cf1849fe66a66bcc2d7b@24.aaa.bbb.7' localhost*CLI> <-- SIP read from 192.168.233.240:5060: SIP/2.0 200 OK Call-ID: 27f7d962409c15d2ac351ca5987858ac@192.168.233.240 CSeq: 106 NOTIFY From: Ira ;tag=as14e35da4 To: 103 ;tag=535790d98069983 Via: SIP/2.0/UDP 192.168.233.235:5060;branch=z9hG4bK54471662 Content-Length: 0 Contact: 103 Supported: replaces User-Agent: Aastra 480i Cordless/1.4.0.1048 Brcm Callctrl/1.5 MxSF/v3.2.6.26 --- (10 headers 0 lines)--- localhost*CLI> <-- SIP read from 192.168.233.237:5060: ACK sip:4323NXXXXXX@192.168.233.235 SIP/2.0 Via: SIP/2.0/UDP 192.168.233.237;branch=z9hG4bKa01e69b84 Max-Forwards: 70 Content-Length: 0 To: 4323NXXXXXX ;tag=as6f853a09 From: 102 ;tag=931535749fef423 Call-ID: c4fafd9f5cbb535473bf4db88cd00c43@192.168.233.237 CSeq: 1148010824 ACK Contact: 102 Proxy-Authorization:Digest response="2961225c154b509a3f51d642d690f4f4",username="102",realm="asterisk",nonce="535a6aef",uri="sip:4323NXXXXXX@192.168.233.235" User-Agent: Aastra 480i Cordless/1.4.0.1048 Brcm Callctrl/1.5 MxSF/v3.2.6.26 --- (11 headers 0 lines)--- set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.233.237, port 5060 Reliably Transmitting (no NAT) to 192.168.233.237:5060: BYE sip:102@192.168.233.237 SIP/2.0 Via: SIP/2.0/UDP 192.168.233.235:5060;branch=z9hG4bK1e044eb5 From: 4323NXXXXXX ;tag=as6f853a09 To: 102 ;tag=931535749fef423 Contact: Call-ID: c4fafd9f5cbb535473bf4db88cd00c43@192.168.233.237 CSeq: 102 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- localhost*CLI> <-- SIP read from 192.168.233.237:5060: SIP/2.0 200 OK Call-ID: c4fafd9f5cbb535473bf4db88cd00c43@192.168.233.237 CSeq: 102 BYE From: 4323NXXXXXX ;tag=as6f853a09 To: 102 ;tag=931535749fef423 Via: SIP/2.0/UDP 192.168.233.235:5060;branch=z9hG4bK1e044eb5 Content-Length: 0 Supported: replaces User-Agent: Aastra 480i Cordless/1.4.0.1048 Brcm Callctrl/1.5 MxSF/v3.2.6.26 --- (9 headers 0 lines)--- Destroying call 'c4fafd9f5cbb535473bf4db88cd00c43@192.168.233.237' localhost*CLI> quit Script done on Sat 03 Jun 2006 02:48:36 PM PDT