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=z9hG4bK87f9d2d62 Max-Forwards: 70 Content-Length: 590 To: 43235899999 From: 102 ;tag=17889c5040048fc Call-ID: ff15095f1fe9079b6e7ff897f3a3bccd@192.168.233.237 CSeq: 775704208 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 409150106 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 - ff15095f1fe9079b6e7ff897f3a3bccd@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 Via: SIP/2.0/UDP 192.168.233.237;branch=z9hG4bK87f9d2d62;received=192.168.233.237 From: 102 ;tag=17889c5040048fc To: 43235899999 ;tag=as18576f9d Call-ID: ff15095f1fe9079b6e7ff897f3a3bccd@192.168.233.237 CSeq: 775704208 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Proxy-Authenticate: Digest realm="asterisk", nonce="1320ffda" Content-Length: 0 --- Scheduling destruction of call 'ff15095f1fe9079b6e7ff897f3a3bccd@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=z9hG4bK87f9d2d62 Max-Forwards: 70 Content-Length: 0 To: 43235899999 ;tag=as18576f9d From: 102 ;tag=17889c5040048fc Call-ID: ff15095f1fe9079b6e7ff897f3a3bccd@192.168.233.237 CSeq: 775704208 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=z9hG4bK2dd1cd118 Max-Forwards: 70 Content-Length: 590 To: 43235899999 From: 102 ;tag=17889c5040048fc Call-ID: ff15095f1fe9079b6e7ff897f3a3bccd@192.168.233.237 CSeq: 775704209 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="e5f42e055cda83d3a3cb2b389783eccf",username="102",realm="asterisk",nonce="1320ffda",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 409150106 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 localhost*CLI> 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 - ff15095f1fe9079b6e7ff897f3a3bccd@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=z9hG4bK2dd1cd118;received=192.168.233.237 From: 102 ;tag=17889c5040048fc To: 43235899999 Call-ID: ff15095f1fe9079b6e7ff897f3a3bccd@192.168.233.237 CSeq: 775704209 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-d4cb", "dial-fish|4|13235899999") in new stack -- Executing NoOp("SIP/102-d4cb", "macro-dial-fish") in new stack -- Executing Set("SIP/102-d4cb", "CALLERID(Name)=FishTuning") in new stack -- Executing Set("SIP/102-d4cb", "CALLERID(Number)=3104990989") in new stack -- Executing Dial("SIP/102-d4cb", "SIP/fishtune/13235899999||j") in new stack -- Called fishtune/13235899999 localhost*CLI> 12 headers, 3 lines Reliably Transmitting (no NAT) to 192.168.233.237:5060: NOTIFY sip:102@192.168.233.237 SIP/2.0 Via: SIP/2.0/UDP 192.168.233.235:5060;branch=z9hG4bK31902d3d From: "asterisk" ;tag=as20910434 To: Contact: Call-ID: 7f42da0c131fa2125c84469c1f582ccd@192.168.233.235 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: message-summary Content-Type: application/simple-message-summary C localhost*CLI> ontent-Length: 95 Messages-Waiting: no Message-Account: sip:asterisk@192.168.233.235 Voice-Message: 0/1 (0/0) --- Scheduling destruction of call '7f42da0c131fa2125c84469c1f582ccd@192.168.233.235' in 15000 ms localhost*CLI> <-- SIP read from 192.168.233.237:5060: SIP/2.0 200 OK Call-ID: 7f42da0c131fa2125c84469c1f582ccd@192.168.233.235 CSeq: 102 NOTIFY From: "asterisk" ;tag=as20910434 To: ;tag=c5bcd51051b9afb Via: SIP/2.0/UDP 192.168.233.235:5060;branch=z9hG4bK31902d3d Content-Length: 0 Contact: 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 '7f42da0c131fa2125c84469c1f582ccd@192.168.233.235' localhost*CLI> -- SIP/fishtune-7254 is making progress passing it to SIP/102-d4cb We're at 192.168.233.235 port 16574 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=z9hG4bK2dd1cd118;received=192.168.233.237 From: 102 ;tag=17889c5040048fc To: 43235899999 ;tag=as1185cd39 Call-ID: ff15095f1fe9079b6e7ff897f3a3bccd@192.168.233.237 CSeq: 775704209 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 16574 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-7254 answered SIP/102-d4cb We're at 192.168.233.235 port 16574 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=z9hG4bK2dd1cd118;received=192.168.233.237 From: 102 ;tag=17889c5040048fc To: 43235899999 ;tag=as1185cd39 Call-ID: ff15095f1fe9079b6e7ff897f3a3bccd@192.168.233.237 localhost*CLI> CSeq: 775704209 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 16574 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-d4cb and SIP/fishtune-7254 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=z9hG4bK98d73171f Max-Forwards: 70 Content-Length: 0 To: 43235899999 ;tag=as1185cd39 From: 102 ;tag=17889c5040048fc Call-ID: ff15095f1fe9079b6e7ff897f3a3bccd@192.168.233.237 CSeq: 775704209 ACK Contact: 102 Proxy-Authorization:Digest response="3e37f72e24dfa89ba619f767ed7f8da5",username="102",realm="asterisk",nonce="1320ffda",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)--- 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=z9hG4bKd5e422a27 Max-Forwards: 70 Content-Length: 0 To: 43235899999 ;tag=as1185cd39 From: 102 ;tag=17889c5040048fc Call-ID: ff15095f1fe9079b6e7ff897f3a3bccd@192.168.233.237 CSeq: 775704210 BYE Supported: timer Supported: replaces Proxy-Authorization:Digest response="1dc370756c63c731c3037df9f11155fb",username="102",realm="asterisk",nonce="1320ffda",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=z9hG4bKd5e422a27;received=192.168.233.237 From: 102 ;tag=17889c5040048fc To: 43235899999 ;tag=as1185cd39 Call-ID: ff15095f1fe9079b6e7ff897f3a3bccd@192.168.233.237 CSeq: 775704210 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> Extension Changed 102 new state Idle for Notify User 103 Extension Changed 102 new state Idle for Notify User 101 localhost*CLI> Destroying call 'ff15095f1fe9079b6e7ff897f3a3bccd@192.168.233.237' localhost*CLI> quit Script done on Fri 02 Jun 2006 06:25:45 PM PDT