=~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2011.11.01 13:13:35 =~=~=~=~=~=~=~=~=~=~=~= asterisk-phone*CLI>  <--- SIP read from UDP:192.168.15.184:5060 ---> INVITE sip:5321@192.168.15.251:5060;user=phone SIP/2.0Via: SIP/2.0/UDP 192.168.15.184;branch=z9hG4bK957ff94859F15BC7From: "WT1" ;tag=E04E80AE-137924ABTo: CSeq: 1 INVITECall-ID: 44959a9a-adc9aa3c-989d5231@192.168.15.184Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFERUser-Agent: PolycomSoundPointIP-SPIP_301-UA/3.1.7.0134Accept-Language: enSupported: 100rel,replacesAllow-Events: talk,hold,conferenceMax-Forwards: 70Content-Type: application/sdpContent-Length: 270v=0o=- 1320167615 1320167615 IN IP4 192.168.15.184s=Polycom IP Phonec=IN IP4 192.168.15.184t=0 0a=sendrecvm=audio 2224 RTP/AVP 0 8 18 101a=rtpmap:0 PCMU/8000a=rtpmap:8 PCMA/8000a=rtpmap:18 G729/8000a=fmtp:18 annexb=noa=rtpmap:101 telephone-event/8000 <-------------> --- (15 headers 12 lines) --- Sending to 192.168.15.184:5060 (no NAT) Using INVITE request as basis request - 44959a9a-adc9aa3c-989d5231@192.168.15.184 Found peer '5261' for '5261' from 192.168.15.184:5060 == Using SIP RTP CoS mark 5 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 101 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format G729 for ID 18 Found audio description format telephone-event for ID 101 Capabilities: us - 0x80e (gsm|ulaw|alaw|g726), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 192.168.15.184:2224 Looking for 5321 in DLPN_LUSI_Internet_Unrestricted (domain 192.168.15.251:5060) asterisk-phone*CLI> list_route: hop: <--- Transmitting (NAT) to 192.168.15.184:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 192.168.15.184;branch=z9hG4bK957ff94859F15BC7;received=192.168.15.184;rport=5060From: "WT1" ;tag=E04E80AE-137924ABTo: Call-ID: 44959a9a-adc9aa3c-989d5231@192.168.15.184CSeq: 1 INVITEServer: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISHSupported: replaces, timerContact: Content-Length: 0 <------------> asterisk-phone*CLI>  -- Executing [5321@DLPN_LUSI_Internet_Unrestricted:1] Dial("SIP/5261-0001b908", "SIP/5321") in new stack asterisk-phone*CLI>  == Using SIP RTP CoS mark 5 asterisk-phone*CLI>  -- Called SIP/5321 asterisk-phone*CLI>  -- SIP/5321-0001b909 is ringing asterisk-phone*CLI>  <--- Transmitting (NAT) to 192.168.15.184:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 192.168.15.184;branch=z9hG4bK957ff94859F15BC7;received=192.168.15.184;rport=5060From: "WT1" ;tag=E04E80AE-137924ABTo: ;tag=as036dbe28Call-ID: 44959a9a-adc9aa3c-989d5231@192.168.15.184CSeq: 1 INVITEServer: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISHSupported: replaces, timerContact: Remote-Party-ID: "Party 1" ;party=called;privacy=off;screen=noContent-Length: 0 <------------> asterisk-phone*CLI> [2011-11-01 13:13:44] WARNING[2629]: chan_sip.c:8738 process_sdp: Unsupported SDP media type in offer: video 0 RTP/AVP 34 98 99 asterisk-phone*CLI>  -- SIP/5321-0001b909 answered SIP/5261-0001b908 Audio is at 5060 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (NAT) to 192.168.15.184:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 192.168.15.184;branch=z9hG4bK957ff94859F15BC7;received=192.168.15.184;rport=5060From: "WT1" ;tag=E04E80AE-137924ABTo: ;tag=as036dbe28Call-ID: 44959a9a-adc9aa3c-989d5231@192.168.15.184CSeq: 1 INVITEServer: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISHSupported: replaces, timerContact: Remote-Party-ID: "Party 1" ;party=called;privacy=off;screen=noContent-Type: application/sdpContent-Length: 260v=0o=root 1343353623 1343353623 IN IP4 192.168.15.251s=Asterisk PBX 1.8.6.0c=IN IP4 192.168.15.251t=0 0m=audio 15908 RTP/AVP 0 8 101a=rtpmap:0 PCMU/8000a=rtpmap:8 PCMA/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=ptime:20a=sendrecv <------------> -- Locally bridging SIP/5261-0001b908 and SIP/5321-0001b909 asterisk-phone*CLI>  <--- SIP read from UDP:192.168.15.184:5060 ---> ACK sip:5321@192.168.15.251:5060 SIP/2.0Via: SIP/2.0/UDP 192.168.15.184;branch=z9hG4bKd44b21b515BDAA70From: "WT1" ;tag=E04E80AE-137924ABTo: ;tag=as036dbe28CSeq: 1 ACKCall-ID: 44959a9a-adc9aa3c-989d5231@192.168.15.184Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFERUser-Agent: PolycomSoundPointIP-SPIP_301-UA/3.1.7.0134Accept-Language: enMax-Forwards: 70Content-Length: 0 <-------------> --- (12 headers 0 lines) --- asterisk-phone*CLI> [2011-11-01 13:13:45] WARNING[2629]: chan_sip.c:8738 process_sdp: Unsupported SDP media type in offer: video 0 RTP/AVP 34 98 99 asterisk-phone*CLI>  -- Started music on hold, class 'default', on SIP/5261-0001b908 asterisk-phone*CLI>  == Using SIP RTP CoS mark 5 asterisk-phone*CLI>  -- Executing [5221@DLPN_DialPlan1:1] Macro("SIP/5321-0001b90a", "stdexten,5221,SIP/5221") in new stack asterisk-phone*CLI>  -- Executing [s@macro-stdexten:1] Set("SIP/5321-0001b90a", "__DYNAMIC_FEATURES=") in new stack asterisk-phone*CLI>  -- Executing [s@macro-stdexten:2] Set("SIP/5321-0001b90a", "ORIG_ARG1=5221") in new stack asterisk-phone*CLI>  -- Executing [s@macro-stdexten:3] GotoIf("SIP/5321-0001b90a", "0?6:4") in new stack asterisk-phone*CLI>  -- Goto (macro-stdexten,s,4) asterisk-phone*CLI>  -- Executing [s@macro-stdexten:4] Dial("SIP/5321-0001b90a", "SIP/5221,30,") in new stack asterisk-phone*CLI>  == Using SIP RTP CoS mark 5 asterisk-phone*CLI>  -- Called SIP/5221 asterisk-phone*CLI>  -- SIP/5221-0001b90b is ringing asterisk-phone*CLI>  -- Stopped music on hold on SIP/5261-0001b908 asterisk-phone*CLI> set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.15.184:5060 Audio is at 5060 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 192.168.15.184:5060: INVITE sip:5261@192.168.15.184 SIP/2.0Via: SIP/2.0/UDP 192.168.15.251:5060;branch=z9hG4bK6c478c42;rportMax-Forwards: 70From: ;tag=as036dbe28To: "WT1" ;tag=E04E80AE-137924ABContact: Call-ID: 44959a9a-adc9aa3c-989d5231@192.168.15.184CSeq: 102 INVITEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISHSupported: replaces, timerRemote-Party-ID: "Party 1" ;party=called;privacy=off;screen=noContent-Type: application/sdpContent-Length: 260v=0o=root 1343353623 1343353624 IN IP4 192.168.15.251s=Asterisk PBX 1.8.6.0c=IN IP4 192.168.15.251t=0 0m=audio 15908 RTP/AVP 0 8 101a=rtpmap:0 PCMU/8000a=rtpmap:8 PCMA/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=ptime:20a=sendrecv --- asterisk-phone*CLI>  -- Got SIP response 500 "Internal Server Error" back from