localhost*CLI> 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=z9hG4bKf5b549393 Max-Forwards: 70 Content-Length: 591 To: 43235899999 From: 102 ;tag=45b12c70c21777f Call-ID: 3c8c9bff3bf022566bc7b2e9a60a4e2d@192.168.233.237 CSeq: 102251346 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 1771440180 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 - 3c8c9bff3bf022566bc7b2e9a60a4e2d@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=z9hG4bKf5b549393;received=192.168.233.237 From: 102 ;tag=45b12c70c21777f To: 43235899999 ;tag=as678c5b25 Call-ID: 3c8c9bff3bf022566bc7b2e9a60a4e2d@192.168.233.237 CSeq: 102251346 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Proxy-Authenticate: Digest realm="asterisk", nonce="41bddfd4" Content-Length: 0 --- Scheduling destruction of call '3c8c9bff3bf022566bc7b2e9a60a4e2d@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=z9hG4bKf5b549393 Max-Forwards: 70 Content-Length: 0 To: 43235899999 ;tag=as678c5b25 From: 102 ;tag=45b12c70c21777f Call-ID: 3c8c9bff3bf022566bc7b2e9a60a4e2d@192.168.233.237 CSeq: 102251346 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=z9hG4bKd7a67204f Max-Forwards: 70 Content-Length: 591 To: 43235899999 From: 102 ;tag=45b12c70c21777f Call-ID: 3c8c9bff3bf022566bc7b2e9a60a4e2d@192.168.233.237 CSeq: 102251347 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="0f80f6bd7514687c3fef3050e9ab1d8d",username="102",realm="asterisk",nonce="41bddfd4",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 1771440180 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 - 3c8c9bff3bf022566bc7b2e9a60a4e2d@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 localhost*CLI> 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=z9hG4bKd7a67204f;received=192.168.233.237 From: 102 ;tag=45b12c70c21777f To: 43235899999 Call-ID: 3c8c9bff3bf022566bc7b2e9a60a4e2d@192.168.233.237 CSeq: 102251347 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- Extension Changed 102 new state InUse for Notify User 103 Extension Changed 102 new state InUse for Notify User 101 localhost*CLI> -- Executing Macro("SIP/102-752b", "dial-fish|4|13235899999") in new stack -- Executing NoOp("SIP/102-752b", "macro-dial-fish") in new stack -- Executing Set("SIP/102-752b", "CALLERID(Name)=FishTuning") in new stack -- Executing Set("SIP/102-752b", "CALLERID(Number)=3104990989") in new stack -- Executing Dial("SIP/102-752b", "SIP/fishtune/13235899999||j") in new stack localhost*CLI> -- Called fishtune/13235899999 localhost*CLI> -- SIP/fishtune-d6da is making progress passing it to SIP/102-752b We're at 192.168.233.235 port 16558 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=z9hG4bKd7a67204f;received=192.168.233.237 From: 102 ;tag=45b12c70c21777f To: 43235899999 ;tag=as71518cc0 Call-ID: 3c8c9bff3bf022566bc7b2e9a60a4e2d@192.168.233.237 CSeq: 102251347 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 6607 6607 IN IP4 192.168.233.235 s=session c=IN IP4 192.168.233.235 t=0 0 m=audio 16558 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/fishtune-d6da answered SIP/102-752b We're at 192.168.233.235 port 16558 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=z9hG4bKd7a67204f;received=192.168.233.237 From: 102 ;tag=45b12c70c21777f To: 43235899999 ;tag=as71518cc0 Call-ID: 3c8c9bff3bf022566bc7b2e9a60a4e2d@192.168.233.237 localhost*CLI> CSeq: 102251347 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 6607 6608 IN IP4 192.168.233.235 s=session c=IN IP4 192.168.233.235 t=0 0 m=audio 16558 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/102-752b and SIP/fishtune-d6da localhost*CLI> Extension Changed 102 new state Idle for Notify User 103 Extension Changed 102 new state Idle for Notify User 101 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=z9hG4bK8bd5a1d6b Max-Forwards: 70 Content-Length: 0 To: 43235899999 ;tag=as71518cc0 From: 102 ;tag=45b12c70c21777f Call-ID: 3c8c9bff3bf022566bc7b2e9a60a4e2d@192.168.233.237 CSeq: 102251347 ACK Contact: 102 Proxy-Authorization:Digest response="a6fcc416975b157dff2f516ecba4faf2",username="102",realm="asterisk",nonce="41bddfd4",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=z9hG4bK2898cad8;rport From: 43235899999 To: 102 ;tag=45b12c70c21777f Contact: Call-ID: 3c8c9bff3bf022566bc7b2e9a60a4e2d@192.168.233.237 CSeq: 101 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Scheduling destruction of call '3c8c9bff3bf022566bc7b2e9a60a4e2d@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=z9hG4bK2898cad8;rport From: 43235899999 To: 102 ;tag=45b12c70c21777f Contact: Call-ID: 3c8c9bff3bf022566bc7b2e9a60a4e2d@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=z9hG4bK2898cad8;rport From: 43235899999 To: 102 ;tag=45b12c70c21777f Contact: Call-ID: 3c8c9bff3bf022566bc7b2e9a60a4e2d@192.168.233.237 CSeq: 101 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- localhost*CLI> Jun 2 18:23:27 NOTICE[6618]: chan_sip.c:5342 sip_reregister: -- Re-registration for XxXxXxX@sip.internetcalls.com localhost*CLI> REGISTER attempt 1 to XxXxXxX@sip.internetcalls.com localhost*CLI> Jun 2 18:23:27 NOTICE[6618]: chan_sip.c:5342 sip_reregister: -- Re-registration for ixXxX@gw4.telasip.com [C localhost*CLI> REGISTER attempt 1 to ixXxX@gw4.telasip.com localhost*CLI> REGISTER attempt 2 to ixXxX@gw4.telasip.com localhost*CLI> Jun 2 18:23:27 NOTICE[6618]: chan_sip.c:5342 sip_reregister: -- Re-registration for XxX_xXxX@sip1.sipdiscount.com localhost*CLI> REGISTER attempt 1 to XxX_xXxX@sip1.sipdiscount.com localhost*CLI> Jun 2 18:23:27 NOTICE[6618]: chan_sip.c:9830 handle_response_register: Outbound Registration: Expiry for gw4.telasip.com is 120 sec (Scheduling reregistration in 105 s) localhost*CLI> Jun 2 18:23:27 NOTICE[6618]: chan_sip.c:9830 handle_response_register: Outbound Registration: Expiry for sip.internetcalls.com is 120 sec (Scheduling reregistration in 105 s) localhost*CLI> Jun 2 18:23:28 NOTICE[6618]: chan_sip.c:9830 handle_response_register: Outbound Registration: Expiry for sip1.sipdiscount.com is 120 sec (Scheduling reregistration in 105 s) 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=z9hG4bK2898cad8;rport From: 43235899999 To: 102 ;tag=45b12c70c21777f Contact: Call-ID: 3c8c9bff3bf022566bc7b2e9a60a4e2d@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=z9hG4bK2898cad8;rport From: 43235899999 To: 102 ;tag=45b12c70c21777f Contact: Call-ID: 3c8c9bff3bf022566bc7b2e9a60a4e2d@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=z9hG4bK2898cad8;rport From: 43235899999 To: 102 ;tag=45b12c70c21777f Contact: Call-ID: 3c8c9bff3bf022566bc7b2e9a60a4e2d@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=z9hG4bKeaafd1506 Max-Forwards: 70 Content-Length: 0 To: 43235899999 ;tag=as71518cc0 From: 102 ;tag=45b12c70c21777f Call-ID: 3c8c9bff3bf022566bc7b2e9a60a4e2d@192.168.233.237 CSeq: 102251348 BYE Supported: timer Supported: replaces Proxy-Authorization:Digest response="910e9e5abf8718087889c6e7c2e7734f",username="102",realm="asterisk",nonce="41bddfd4",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=z9hG4bKeaafd1506;received=192.168.233.237 From: 102 ;tag=45b12c70c21777f To: 43235899999 ;tag=as71518cc0 Call-ID: 3c8c9bff3bf022566bc7b2e9a60a4e2d@192.168.233.237 CSeq: 102251348 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: 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=z9hG4bK2898cad8;rport From: 43235899999 To: 102 ;tag=45b12c70c21777f Contact: Call-ID: 3c8c9bff3bf022566bc7b2e9a60a4e2d@192.168.233.237 CSeq: 101 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- localhost*CLI> 12 headers, 0 lines Reliably Transmitting (no NAT) to 192.168.233.237:5060: OPTIONS sip:102@192.168.233.237 SIP/2.0 Via: SIP/2.0/UDP 192.168.233.235:5060;branch=z9hG4bK3f1632e3 From: "asterisk" ;tag=as2ef3213e To: Contact: Call-ID: 0c12dd5403422b3a4a0c3ecf25116940@192.168.233.235 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Sat, 03 Jun 2006 01:23:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- localhost*CLI> <-- SIP read from 192.168.233.237:5060: SIP/2.0 200 OK Call-ID: 0c12dd5403422b3a4a0c3ecf25116940@192.168.233.235 CSeq: 102 OPTIONS From: "asterisk" ;tag=as2ef3213e To: ;tag=41a0435fcf1327f Via: SIP/2.0/UDP 192.168.233.235:5060;branch=z9hG4bK3f1632e3 Content-Length: 0 Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO Contact: Supported: replaces User-Agent: Aastra 480i Cordless/1.4.0.1048 Brcm Callctrl/1.5 MxSF/v3.2.6.26 --- (11 headers 0 lines)--- Destroying call '0c12dd5403422b3a4a0c3ecf25116940@192.168.233.235' localhost*CLI> Jun 2 18:23:44 WARNING[6618]: chan_sip.c:1217 retrans_pkt: Maximum retries exceeded on transmission 3c8c9bff3bf022566bc7b2e9a60a4e2d@192.168.233.237 for seqno 101 (Non-critical Request) Destroying call '3c8c9bff3bf022566bc7b2e9a60a4e2d@192.168.233.237' 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=z9hG4bKe409e4008 Max-Forwards: 70 Content-Length: 590 To: 43235899999 From: 102 ;tag=bcb93780dab9cf4 Call-ID: 608c228dcefe9ab249b62e3304f5dde9@192.168.233.237 CSeq: 1100678766 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 928767281 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 - 608c228dcefe9ab249b62e3304f5dde9@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=z9hG4bKe409e4008;received=192.168.233.237 From: 102 ;tag=bcb93780dab9cf4 To: 43235899999 ;tag=as6d9b6ca7 Call-ID: 608c228dcefe9ab249b62e3304f5dde9@192.168.233.237 CSeq: 1100678766 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Proxy-Authenticate: Digest realm="asterisk", nonce="5db5fee8" Content-Length: 0 --- Scheduling destruction of call '608c228dcefe9ab249b62e3304f5dde9@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=z9hG4bKe409e4008 Max-Forwards: 70 Content-Length: 0 To: 43235899999 ;tag=as6d9b6ca7 From: 102 ;tag=bcb93780dab9cf4 Call-ID: 608c228dcefe9ab249b62e3304f5dde9@192.168.233.237 CSeq: 1100678766 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=z9hG4bK958ecac0d Max-Forwards: 70 Content-Length: 590 To: 43235899999 From: 102 ;tag=bcb93780dab9cf4 Call-ID: 608c228dcefe9ab249b62e3304f5dde9@192.168.233.237 CSeq: 1100678767 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="1e0066fb1d133040ace4a26786340e82",username="102",realm="asterisk",nonce="5db5fee8",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 928767281 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 - 608c228dcefe9ab249b62e3304f5dde9@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 localhost*CLI> 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=z9hG4bK958ecac0d;received=192.168.233.237 From: 102 ;tag=bcb93780dab9cf4 To: 43235899999 Call-ID: 608c228dcefe9ab249b62e3304f5dde9@192.168.233.237 CSeq: 1100678767 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- Extension Changed 102 new state InUse for Notify User 103 Extension Changed 102 new state InUse for Notify User 101 -- Executing Macro("SIP/102-11de", "dial-fish|4|13235899999") in new stack -- Executing NoOp("SIP/102-11de", "macro-dial-fish") in new stack -- Executing Set("SIP/102-11de", "CALLERID(Name)=FishTuning") in new stack -- Executing Set("SIP/102-11de", "CALLERID(Number)=3104990989") in new stack -- Executing Dial("SIP/102-11de", "SIP/fishtune/13235899999||j") in new stack -- Called fishtune/13235899999 localhost*CLI> -- SIP/fishtune-dff3 is making progress passing it to SIP/102-11de We're at 192.168.233.235 port 16266 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=z9hG4bK958ecac0d;received=192.168.233.237 From: 102 ;tag=bcb93780dab9cf4 To: 43235899999 ;tag=as2dfb6e5c Call-ID: 608c228dcefe9ab249b62e3304f5dde9@192.168.233.237 CSeq: 1100678767 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 6607 6607 IN IP4 192.168.233.235 s=session c=IN IP4 192.168.233.235 t=0 0 m=audio 16266 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/fishtune-dff3 answered SIP/102-11de We're at 192.168.233.235 port 16266 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=z9hG4bK958ecac0d;received=192.168.233.237 From: 102 ;tag=bcb93780dab9cf4 To: 43235899999 ;tag=as2dfb6e5c Call-ID: 608c228dcefe9ab249b62e3304f5dde9@192.168.233.237 localhost*CLI> CSeq: 1100678767 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 6607 6608 IN IP4 192.168.233.235 s=session c=IN IP4 192.168.233.235 t=0 0 m=audio 16266 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/102-11de and SIP/fishtune-dff3 localhost*CLI> Extension Changed 102 new state Idle for Notify User 103 Extension Changed 102 new state Idle for Notify User 101 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=z9hG4bK05c537e89 Max-Forwards: 70 Content-Length: 0 To: 43235899999 ;tag=as2dfb6e5c From: 102 ;tag=bcb93780dab9cf4 Call-ID: 608c228dcefe9ab249b62e3304f5dde9@192.168.233.237 CSeq: 1100678767 ACK Contact: 102 Proxy-Authorization:Digest response="a4653111c1e6f1b7f1b0310115e7cd53",username="102",realm="asterisk",nonce="5db5fee8",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=z9hG4bK567a8f23;rport From: 43235899999 To: 102 ;tag=bcb93780dab9cf4 Contact: Call-ID: 608c228dcefe9ab249b62e3304f5dde9@192.168.233.237 CSeq: 101 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Scheduling destruction of call '608c228dcefe9ab249b62e3304f5dde9@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=z9hG4bK567a8f23;rport From: 43235899999 To: 102 ;tag=bcb93780dab9cf4 Contact: Call-ID: 608c228dcefe9ab249b62e3304f5dde9@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=z9hG4bK567a8f23;rport From: 43235899999 To: 102 ;tag=bcb93780dab9cf4 Contact: Call-ID: 608c228dcefe9ab249b62e3304f5dde9@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=z9hG4bK567a8f23;rport From: 43235899999 To: 102 ;tag=bcb93780dab9cf4 Contact: Call-ID: 608c228dcefe9ab249b62e3304f5dde9@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=z9hG4bK567a8f23;rport From: 43235899999 To: 102 ;tag=bcb93780dab9cf4 Contact: Call-ID: 608c228dcefe9ab249b62e3304f5dde9@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=z9hG4bK567a8f23;rport From: 43235899999 To: 102 ;tag=bcb93780dab9cf4 Contact: Call-ID: 608c228dcefe9ab249b62e3304f5dde9@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=z9hG4bK456e28c37 Max-Forwards: 70 Content-Length: 0 To: 43235899999 ;tag=as2dfb6e5c From: 102 ;tag=bcb93780dab9cf4 Call-ID: 608c228dcefe9ab249b62e3304f5dde9@192.168.233.237 CSeq: 1100678768 BYE Supported: timer Supported: replaces Proxy-Authorization:Digest response="74671f36162535f882b4869fd996bef9",username="102",realm="asterisk",nonce="5db5fee8",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=z9hG4bK456e28c37;received=192.168.233.237 From: 102 ;tag=bcb93780dab9cf4 To: 43235899999 ;tag=as2dfb6e5c Call-ID: 608c228dcefe9ab249b62e3304f5dde9@192.168.233.237 CSeq: 1100678768 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: 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=z9hG4bK567a8f23;rport From: 43235899999 To: 102 ;tag=bcb93780dab9cf4 Contact: Call-ID: 608c228dcefe9ab249b62e3304f5dde9@192.168.233.237 CSeq: 101 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- localhost*CLI> Jun 2 18:24:22 WARNING[6618]: chan_sip.c:1217 retrans_pkt: Maximum retries exceeded on transmission 608c228dcefe9ab249b62e3304f5dde9@192.168.233.237 for seqno 101 (Non-critical Request) Destroying call '608c228dcefe9ab249b62e3304f5dde9@192.168.233.237' localhost*CLI> quit Script done on Fri 02 Jun 2006 06:24:24 PM PDT