Script started on Thu 01 Jun 2006 01:11:20 PM PDT [root@localhost asterisk]# exec asterisk -r Asterisk 1.2.8, 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 1.2.8 currently running on localhost (pid = 5647) localhost*CLI> Verbosity is at least 16 localhost*CLI> sip debug localhost*CLI> SIP Debugging enabled 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=z9hG4bKc29cb1241 Max-Forwards: 70 Content-Length: 591 To: 43235899999 From: 102 ;tag=5239641ab3f4e81 Call-ID: 315ba2ab92600671790530aa6c14f334@192.168.233.237 CSeq: 1508874154 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 1305049213 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 - 315ba2ab92600671790530aa6c14f334@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 localhost*CLI> Via: SIP/2.0/UDP 192.168.233.237;branch=z9hG4bKc29cb1241;received=192.168.233.237 From: 102 ;tag=5239641ab3f4e81 To: 43235899999 ;tag=as02a7b00a Call-ID: 315ba2ab92600671790530aa6c14f334@192.168.233.237 CSeq: 1508874154 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Proxy-Authenticate: Digest realm="asterisk", nonce="0a220eac" Content-Length: 0 --- Scheduling destruction of call '315ba2ab92600671790530aa6c14f334@192.168.233.237' in 15000 ms Found user '102' localhost*CLI> <-- SIP read from 192.168.233.237:5060: ACK sip:43235899999@192.168.233.235:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.233.237;branch=z9hG4bKc29cb1241 Max-Forwards: 70 Content-Length: 0 To: 43235899999 ;tag=as02a7b00a From: 102 ;tag=5239641ab3f4e81 Call-ID: 315ba2ab92600671790530aa6c14f334@192.168.233.237 CSeq: 1508874154 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:43235899999@192.168.233.235:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.233.237;branch=z9hG4bK59b8d8af9 Max-Forwards: 70 Content-Length: 591 To: 43235899999 From: 102 ;tag=5239641ab3f4e81 Call-ID: 315ba2ab92600671790530aa6c14f334@192.168.233.237 CSeq: 1508874155 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="1f26998d96fb5999d0eeb8242c28c39f",username="102",realm="asterisk",nonce="0a220eac",uri="sip:43235899999@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 1305049213 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 - 315ba2ab92600671790530aa6c14f334@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 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 43235899999 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=z9hG4bK59b8d8af9;received=192.168.233.237 From: 102 ;tag=5239641ab3f4e81 To: 43235899999 Call-ID: 315ba2ab92600671790530aa6c14f334@192.168.233.237 CSeq: 1508874155 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=z9hG4bK4cf92373 From: Ira ;tag=as14e35da4 To: 103 ;tag=535790d98069983 Contact: Call-ID: 27f7d962409c15d2ac351ca5987858ac@192.168.233.240 CSeq: 158 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 211 confirmed --- 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=z9hG4bK7cf3e5fb From: Ira ;tag=as6e4a139b To: 101 ;tag=57883542f06cd06 Contact: Call-ID: 5b30f7b60c1b618505c2d4b30159808b@192.168.233.233 CSeq: 157 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 211 confirmed --- -- Executing Macro("SIP/102-1284", "dial-fish|4|13235899999") in new stack -- Executing Dial("SIP/102-1284", "SIP/fishtune/13235899999||j") in new stack We're at MY.IP.ADD.RES port 16190 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP 13 headers, 10 lines Reliably Transmitting (NAT) to MY.SIP.PROVIDER.IP:5060: INVITE sip:13235899999@gw4.MySIPprovider.com SIP/2.0 Via: SIP/2.0/UDP MY.IP.ADD.RES:5060;branch=z9hG4bK2556ba07;rport From: "102" ;tag=as105a6183 To: Contact: Call-ID: 7fe624e3113986cf17b27ac46ab02b21@MY.IP.ADD.RES CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 01 Jun 2006 20:11:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 214 v=0 o=root 5647 5647 IN IP4 MY.IP.ADD.RES s=session c=IN IP4 MY.IP.ADD.RES t=0 0 m=audio 16190 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Called fishtune/13235899999 localhost*CLI> <-- SIP read from 192.168.233.233:5060: SIP/2.0 200 OK Call-ID: 5b30f7b60c1b618505c2d4b30159808b@192.168.233.233 CSeq: 157 NOTIFY From: Ira ;tag=as6e4a139b To: 101 ;tag=57883542f06cd06 Via: SIP/2.0/UDP 192.168.233.235:5060;branch=z9hG4bK7cf3e5fb 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)--- localhost*CLI> <-- SIP read from 192.168.233.240:5060: SIP/2.0 200 OK Call-ID: 27f7d962409c15d2ac351ca5987858ac@192.168.233.240 CSeq: 158 NOTIFY From: Ira ;tag=as14e35da4 To: 103 ;tag=535790d98069983 Via: SIP/2.0/UDP 192.168.233.235:5060;branch=z9hG4bK4cf92373 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 MY.SIP.PROVIDER.IP:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP MY.IP.ADD.RES:5060;branch=z9hG4bK2556ba07;received=MY.IP.ADD.RES;rport=5060 From: "102" ;tag=as105a6183 To: ;tag=as751a5255 Call-ID: 7fe624e3113986cf17b27ac46ab02b21@MY.IP.ADD.RES CSeq: 102 INVITE User-Agent: MySIPprovider Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Proxy-Authenticate: Digest realm="MySIPprovider.com", nonce="35cb1e01" Content-Length: 0 --- (11 headers 0 lines)--- Transmitting (NAT) to MY.SIP.PROVIDER.IP:5060: ACK sip:13235899999@gw4.MySIPprovider.com SIP/2.0 Via: SIP/2.0/UDP MY.IP.ADD.RES:5060;branch=z9hG4bK2556ba07;rport From: "102" ;tag=as105a6183 To: ;tag=as751a5255 Contact: Call-ID: 7fe624e3113986cf17b27ac46ab02b21@MY.IP.ADD.RES CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 localhost*CLI> --- We're at MY.IP.ADD.RES port 16190 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to MY.SIP.PROVIDER.IP:5060: INVITE sip:13235899999@gw4.MySIPprovider.com SIP/2.0 Via: SIP/2.0/UDP MY.IP.ADD.RES:5060;branch=z9hG4bK6e9c4e47;rport From: "102" ;tag=as105a6183 To: Contact: Call-ID: 7fe624e3113986cf17b27ac46ab02b21@MY.IP.ADD.RES CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Proxy-Authorization: Digest username="MyUserName", realm="MySIPprovider.com", algorithm=MD5, uri="sip:13235899999@gw4.MySIPprovider.com", nonce="35cb1e01", response="8f1e859db8303b8b2218f710a4ea660e", opaque="" Date: Thu, 01 Jun 2006 20:11:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 214 v=0 o=root 5647 5648 IN IP4 MY.IP.ADD.RES s=session c=IN IP4 MY.IP.ADD.RES t=0 0 m=audio 16190 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 MY.SIP.PROVIDER.IP:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP MY.IP.ADD.RES:5060;branch=z9hG4bK6e9c4e47;received=MY.IP.ADD.RES;rport=5060 From: "102" ;tag=as105a6183 To: Call-ID: 7fe624e3113986cf17b27ac46ab02b21@MY.IP.ADD.RES CSeq: 103 INVITE User-Agent: MySIPprovider Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- (10 headers 0 lines)--- localhost*CLI> <-- SIP read from MY.SIP.PROVIDER.IP:5060: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP MY.IP.ADD.RES:5060;branch=z9hG4bK6e9c4e47;received=MY.IP.ADD.RES;rport=5060 From: "102" ;tag=as105a6183 To: ;tag=as2d3c2dce Call-ID: 7fe624e3113986cf17b27ac46ab02b21@MY.IP.ADD.RES CSeq: 103 INVITE User-Agent: MySIPprovider Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 210 v=0 o=root 8492 8492 IN IP4 MY.SIP.PROVIDER.IP s=session c=IN IP4 MY.SIP.PROVIDER.IP t=0 0 m=audio 12302 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 MY.SIP.PROVIDER.IP:12302 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-4a59 is making progress passing it to SIP/102-1284 We're at 192.168.233.235 port 16200 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=z9hG4bK59b8d8af9;received=192.168.233.237 From: 102 ;tag=5239641ab3f4e81 To: 43235899999 ;tag=as56645c87 Call-ID: 315ba2ab92600671790530aa6c14f334@192.168.233.237 CSeq: 1508874155 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 5647 5647 IN IP4 192.168.233.235 s=session c=IN IP4 192.168.233.235 t=0 0 m=audio 16200 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 MY.SIP.PROVIDER.IP:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP MY.IP.ADD.RES:5060;branch=z9hG4bK6e9c4e47;received=MY.IP.ADD.RES;rport=5060 From: "102" ;tag=as105a6183 To: ;tag=as2d3c2dce Call-ID: 7fe624e3113986cf17b27ac46ab02b21@MY.IP.ADD.RES CSeq: 103 INVITE User-Agent: MySIPprovider Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 210 v=0 localhost*CLI> o=root 8492 8493 IN IP4 MY.SIP.PROVIDER.IP s=session c=IN IP4 MY.SIP.PROVIDER.IP t=0 0 m=audio 12302 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 MY.SIP.PROVIDER.IP:12302 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 MY.SIP.PROVIDER.IP, port 5060 Transmitting (NAT) to MY.SIP.PROVIDER.IP:5060: ACK sip:13235899999@MY.SIP.PROVIDER.IP SIP/2.0 Via: SIP/2.0/UDP MY.IP.ADD.RES:5060;branch=z9hG4bK3ac5c1ef;rport From: "102" ;tag=as105a6183 To: ;tag=as2d3c2dce Contact: Call-ID: 7fe624e3113986cf17b27ac46ab02b21@MY.IP.ADD.RES CSeq: 103 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- SIP/fishtune-4a59 answered SIP/102-1284 We're at 192.168.233.235 port 16200 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=z9hG4bK59b8d8af9;received=192.168.233.237 From: 102 ;tag=5239641ab3f4e81 To: 43235899999 ;tag=as56645c87 Call-ID: 315ba2ab92600671790530aa6c14f334@192.168.233.237 CSeq: 1508874155 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 5647 5648 IN IP4 192.168.233.235 s=session c=IN IP4 192.168.233.235 t=0 0 m=audio 16200 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/102-1284 and SIP/fishtune-4a59 localhost*CLI> <-- SIP read from MY.SIP.PROVIDER.IP:5060: BYE sip:102@MY.IP.ADD.RES:5060 SIP/2.0 Via: SIP/2.0/UDP MY.SIP.PROVIDER.IP:5060;branch=z9hG4bK17b371cb;rport From: ;tag=as2d3c2dce To: "102" ;tag=as105a6183 Contact: Call-ID: 7fe624e3113986cf17b27ac46ab02b21@MY.IP.ADD.RES CSeq: 102 BYE User-Agent: MySIPprovider Max-Forwards: 70 Content-Length: 0 --- (10 headers 0 lines)--- Sending to MY.SIP.PROVIDER.IP : 5060 (NAT) Transmitting (NAT) to MY.SIP.PROVIDER.IP:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP MY.SIP.PROVIDER.IP:5060;branch=z9hG4bK17b371cb;received=MY.SIP.PROVIDER.IP;rport=5060 From: ;tag=as2d3c2dce To: "102" ;tag=as105a6183 Call-ID: 7fe624e3113986cf17b27ac46ab02b21@MY.IP.ADD.RES 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 --- == Spawn extension (macro-dial-fish, s, 1) exited non-zero on 'SIP/102-1284' in macro 'dial-fish' == Spawn extension (macro-dial-fish, s, 1) exited non-zero on 'SIP/102-1284' 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=z9hG4bK49705282 From: Ira ;tag=as14e35da4 To: 103 ;tag=535790d98069983 Contact: Call-ID: 27f7d962409c15d2ac351ca5987858ac@192.168.233.240 CSeq: 159 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 212 terminated --- 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=z9hG4bK01d19483 From: Ira ;tag=as6e4a139b To: 101 ;tag=57883542f06cd06 Contact: Call-ID: 5b30f7b60c1b618505c2d4b30159808b@192.168.233.233 CSeq: 158 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 212 terminated --- localhost*CLI> <-- SIP read from 192.168.233.233:5060: SIP/2.0 200 OK Call-ID: 5b30f7b60c1b618505c2d4b30159808b@192.168.233.233 CSeq: 158 NOTIFY From: Ira ;tag=as6e4a139b To: 101 ;tag=57883542f06cd06 Via: SIP/2.0/UDP 192.168.233.235:5060;branch=z9hG4bK01d19483 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 '7fe624e3113986cf17b27ac46ab02b21@MY.IP.ADD.RES' localhost*CLI> <-- SIP read from 192.168.233.240:5060: SIP/2.0 200 OK Call-ID: 27f7d962409c15d2ac351ca5987858ac@192.168.233.240 CSeq: 159 NOTIFY From: Ira ;tag=as14e35da4 To: 103 ;tag=535790d98069983 Via: SIP/2.0/UDP 192.168.233.235:5060;branch=z9hG4bK49705282 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:43235899999@192.168.233.235 SIP/2.0 Via: SIP/2.0/UDP 192.168.233.237;branch=z9hG4bK333dd90d4 Max-Forwards: 70 Content-Length: 0 To: 43235899999 ;tag=as56645c87 From: 102 ;tag=5239641ab3f4e81 Call-ID: 315ba2ab92600671790530aa6c14f334@192.168.233.237 CSeq: 1508874155 ACK Contact: 102 Proxy-Authorization:Digest response="693dbe303c400469807cbaf369b56333",username="102",realm="asterisk",nonce="0a220eac",uri="sip:43235899999@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: CANCEL :43235899999@192.168.233.235 SIP/2.0 Via: SIP/2.0/UDP 192.168.233.235:5060;branch=z9hG4bK093d3f3c;rport From: 43235899999 To: 102 ;tag=5239641ab3f4e81 Contact: Call-ID: 315ba2ab92600671790530aa6c14f334@192.168.233.237 CSeq: 101 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Scheduling destruction of call '315ba2ab92600671790530aa6c14f334@192.168.233.237' in 32000 ms localhost*CLI> Retransmitting #1 (no NAT) to 192.168.233.237:5060: CANCEL :43235899999@192.168.233.235 SIP/2.0 Via: SIP/2.0/UDP 192.168.233.235:5060;branch=z9hG4bK093d3f3c;rport From: 43235899999 To: 102 ;tag=5239641ab3f4e81 Contact: Call-ID: 315ba2ab92600671790530aa6c14f334@192.168.233.237 CSeq: 101 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- localhost*CLI> Retransmitting #2 (no NAT) to 192.168.233.237:5060: CANCEL :43235899999@192.168.233.235 SIP/2.0 Via: SIP/2.0/UDP 192.168.233.235:5060;branch=z9hG4bK093d3f3c;rport From: 43235899999 To: 102 ;tag=5239641ab3f4e81 Contact: Call-ID: 315ba2ab92600671790530aa6c14f334@192.168.233.237 CSeq: 101 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- localhost*CLI> Retransmitting #3 (no NAT) to 192.168.233.237:5060: CANCEL :43235899999@192.168.233.235 SIP/2.0 Via: SIP/2.0/UDP 192.168.233.235:5060;branch=z9hG4bK093d3f3c;rport From: 43235899999 To: 102 ;tag=5239641ab3f4e81 Contact: Call-ID: 315ba2ab92600671790530aa6c14f334@192.168.233.237 CSeq: 101 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- localhost*CLI> Retransmitting #4 (no NAT) to 192.168.233.237:5060: CANCEL :43235899999@192.168.233.235 SIP/2.0 Via: SIP/2.0/UDP 192.168.233.235:5060;branch=z9hG4bK093d3f3c;rport From: 43235899999 To: 102 ;tag=5239641ab3f4e81 Contact: Call-ID: 315ba2ab92600671790530aa6c14f334@192.168.233.237 CSeq: 101 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- localhost*CLI> Retransmitting #5 (no NAT) to 192.168.233.237:5060: CANCEL :43235899999@192.168.233.235 SIP/2.0 Via: SIP/2.0/UDP 192.168.233.235:5060;branch=z9hG4bK093d3f3c;rport From: 43235899999 To: 102 ;tag=5239641ab3f4e81 Contact: Call-ID: 315ba2ab92600671790530aa6c14f334@192.168.233.237 CSeq: 101 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- localhost*CLI> Retransmitting #6 (no NAT) to 192.168.233.237:5060: CANCEL :43235899999@192.168.233.235 SIP/2.0 Via: SIP/2.0/UDP 192.168.233.235:5060;branch=z9hG4bK093d3f3c;rport From: 43235899999 To: 102 ;tag=5239641ab3f4e81 Contact: Call-ID: 315ba2ab92600671790530aa6c14f334@192.168.233.237 CSeq: 101 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- localhost*CLI> <-- SIP read from 192.168.233.237:5060: BYE sip:43235899999@192.168.233.235 SIP/2.0 Via: SIP/2.0/UDP 192.168.233.237;branch=z9hG4bK22678040e Max-Forwards: 70 Content-Length: 0 To: 43235899999 ;tag=as56645c87 From: 102 ;tag=5239641ab3f4e81 Call-ID: 315ba2ab92600671790530aa6c14f334@192.168.233.237 CSeq: 1508874156 BYE Supported: timer Supported: replaces Proxy-Authorization:Digest response="ddce553655652bbae4d0b855b867988f",username="102",realm="asterisk",nonce="0a220eac",uri="sip:43235899999@192.168.233.235" User-Agent: Aastra 480i Cordless/1.4.0.1048 Brcm Callctrl/1.5 MxSF/v3.2.6.26 --- (12 headers 0 lines)--- Sending to 192.168.233.237 : 5060 (non-NAT) Transmitting (no NAT) to 192.168.233.237:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.233.237;branch=z9hG4bK22678040e;received=192.168.233.237 From: 102 ;tag=5239641ab3f4e81 To: 43235899999 ;tag=as56645c87 Call-ID: 315ba2ab92600671790530aa6c14f334@192.168.233.237 CSeq: 1508874156 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- localhost*CLI> Jun 1 13:11:59 WARNING[5658]: chan_sip.c:1217 retrans_pkt: Maximum retries exceeded on transmission 315ba2ab92600671790530aa6c14f334@192.168.233.237 for seqno 101 (Non-critical Request) Destroying call '315ba2ab92600671790530aa6c14f334@192.168.233.237' localhost*CLI> quit Script done on Thu 01 Jun 2006 01:12:02 PM PDT