=~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2006.12.01 09:50:05 =~=~=~=~=~=~=~=~=~=~=~= <--- SIP read from 208.51.25.114:5060 ---> OPTIONS sip:89.202.128.200:5060 SIP/2.0 Via: SIP/2.0/UDP 208.51.25.114:5060;branch=z9hG4bK0go4ji20a0pgihgk93g1.1 Call-ID: 6e8f7eb39ab9b7d471747d5dd4a76662@208.51.25.114 To: sip:ping@89.202.128.200 From: sip:ping@208.51.25.114;tag=108352804eb67c3de2e195c22418591f Max-Forwards: 70 CSeq: 12872 OPTIONS <-------------> --- (7 headers 0 lines) --- Looking for s in default (domain 89.202.128.200) <--- Transmitting (no NAT) to 208.51.25.114:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 208.51.25.114:5060;branch=z9hG4bK0go4ji20a0pgihgk93g1.1;received=208.51.25.114 From: sip:ping@208.51.25.114;tag=108352804eb67c3de2e195c22418591f To: sip:ping@89.202.128.200;tag=as458949ae Call-ID: 6e8f7eb39ab9b7d471747d5dd4a76662@208.51.25.114 CSeq: 12872 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Accept: application/sdp Content-Length: 0 <------------> Scheduling destruction of SIP dialog '6e8f7eb39ab9b7d471747d5dd4a76662@208.51.25.114' in 32000 ms (Method: OPTIONS) m900a*CLI> <--- SIP read from 83.245.6.83:5060 ---> INVITE sip:448001180118@89.202.128.200 SIP/2.0 Max-Forwards: 11 Session-Expires: 3600;Refresher=uac Supported: timer To: From: ;tag=3373955424-319415 Remote-Party-Id: ;party=calling;screen=yes;privacy=off Call-ID: 1518484-3373955424-319378@msw8.mydomain.com CSeq: 1 INVITE Via: SIP/2.0/UDP 83.245.6.83:5060;branch=z9hG4bKff453fd2d4ffb4804d02ab3e894c16a3 Contact: sip:442074904605@83.245.6.83:5060 Allow-Events: telephone-event Content-Type: application/sdp Content-Length: 246 v=0 o=msw 3593 7553 IN IP4 83.245.6.83 s=sip call c=IN IP4 83.245.6.84 t=0 0 m=audio 62298 RTP/AVP 0 100 101 a=ptime:20 a=fmtp:101 0-16 a=rtpmap:101 telephone-event/8000 a=fmtp:100 192-194 a=rtpmap:100 X-NSE/8000 a=rtpmap:0 PCMU/8000 <-------------> --- (14 headers 12 lines) --- Sending to 83.245.6.83 : 5060 (no NAT) Using INVITE request as basis request - 1518484-3373955424-319378@msw8.mydomain.com No user '442074904605' in SIP users list Found peer 'gamma-OUT' for '442074904605' from 83.245.6.83:5060 Found RTP audio format 0 Found RTP audio format 100 Found RTP audio format 101 Peer audio RTP is at port 83.245.6.84:62298 Got unsupported a:fmtp in SDP offer Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Found description format X-NSE for ID 100 Found description format PCMU for ID 0 Capabilities: us - 0x3f1fff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|g722|jpeg|png|h261|h263|h263p|h264), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 83.245.6.84:62298 Looking for 448001180118 in from-gamma (domain 89.202.128.200) list_route: hop: <--- Transmitting (no NAT) to 83.245.6.83:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 83.245.6.83:5060;branch=z9hG4bKff453fd2d4ffb4804d02ab3e894c16a3;received=83.245.6.83 From: ;tag=3373955424-319415 To: Call-ID: 1518484-3373955424-319378@msw8.mydomain.com CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> m900a*CLI> -- Executing [448001180118@from-gamma:1] DeadAGI("SIP/83.245.6.83-0828a348", "agi://127.0.0.1:4574/numbertrans?number_in=448001180118&context=gamma") in new stack m900a*CLI> -- AGI Script Executing Application: (Dial) Options: (Local/442070605610@gamma/n) -- Called 442070605610@gamma/n m900a*CLI> -- Executing [442070605610@gamma:1] Set("Local/442070605610@gamma-c7a2,2", "CALLERID(all)=442074904605") in new stack m900a*CLI> -- Executing [442070605610@gamma:2] Set("Local/442070605610@gamma-c7a2,2", "CALLERID(ANI)=123") in new stack m900a*CLI> -- Executing [442070605610@gamma:3] SetCallerPres("Local/442070605610@gamma-c7a2,2", "allowed") in new stack m900a*CLI> -- Executing [442070605610@gamma:4] Dial("Local/442070605610@gamma-c7a2,2", "SIP/gamma-OUT/442070605610") in new stack m900a*CLI> Audio is at 89.202.128.200 port 15692 m900a*CLI> Adding codec 0x4 (ulaw) to SDP m900a*CLI> Adding codec 0x8 (alaw) to SDP m900a*CLI> Adding codec 0x2 (gsm) to SDP m900a*CLI> Adding codec 0x10 (g726aal2) to SDP m900a*CLI> Adding codec 0x20 (adpcm) to SDP m900a*CLI> Adding codec 0x40 (slin) to SDP m900a*CLI> Adding codec 0x80 (lpc10) to SDP m900a*CLI> Adding codec 0x400 (ilbc) to SDP m900a*CLI> Adding codec 0x800 (g726) to SDP m900a*CLI> Adding non-codec 0x1 (telephone-event) to SDP m900a*CLI> Reliably Transmitting (no NAT) to 83.245.6.83:5060: INVITE sip:442070605610@83.245.6.83 SIP/2.0 Via: SIP/2.0/UDP 89.202.128.200:5060;branch=z9hG4bK07afb6d9;rport Max-Forwards: 70 From: "442074904605" ;tag=as08373072 To: Contact: Call-ID: 5da9f87d1004dd9e31c9e257573f4261@89.202.128.200 CSeq: 102 INVITE User-Agent: Asterisk PBX Remote-Party-ID: "442074904605" ;privacy=off;screen=yes Date: Fri, 01 Dec 2006 09:49:37 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 475 v=0 o=root 20874 20874 IN IP4 89.202.128.200 s=session c=IN IP4 89.202.128.200 t=0 0 m=audio 15692 RTP/AVP 0 8 3 112 5 10 7 97 111 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:112 AAL2-G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:7 LPC/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:111 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- m900a*CLI> -- Called gamma-OUT/442070605610 m900a*CLI> <--- SIP read from 83.245.6.83:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 89.202.128.200:5060;branch=z9hG4bK07afb6d9;rport From: "442074904605" ;tag=as08373072 To: Call-ID: 5da9f87d1004dd9e31c9e257573f4261@89.202.128.200 CSeq: 102 INVITE Content-Length: 0 <-------------> --- (7 headers 0 lines) --- m900a*CLI> <--- SIP read from 208.51.25.114:5060 ---> OPTIONS sip:89.202.128.200:5060 SIP/2.0 Via: SIP/2.0/UDP 208.51.25.114:5060;branch=z9hG4bK0go43l20a0pgihouv4o1.1 Call-ID: 6e8f7eb39ab9b7d471747d5dd4a76662@208.51.25.114 To: sip:ping@89.202.128.200 From: sip:ping@208.51.25.114;tag=108352804eb67c3de2e195c22418591f Max-Forwards: 70 CSeq: 12873 OPTIONS <-------------> --- (7 headers 0 lines) --- Looking for s in default (domain 89.202.128.200) <--- Transmitting (no NAT) to 208.51.25.114:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 208.51.25.114:5060;branch=z9hG4bK0go43l20a0pgihouv4o1.1;received=208.51.25.114 From: sip:ping@208.51.25.114;tag=108352804eb67c3de2e195c22418591f To: sip:ping@89.202.128.200;tag=as458949ae Call-ID: 6e8f7eb39ab9b7d471747d5dd4a76662@208.51.25.114 CSeq: 12873 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Accept: application/sdp Content-Length: 0 <------------> Scheduling destruction of SIP dialog '6e8f7eb39ab9b7d471747d5dd4a76662@208.51.25.114' in 32000 ms (Method: OPTIONS) m900a*CLI> <--- SIP read from 83.245.6.83:5060 ---> SIP/2.0 180 Ringing To: ;tag=3373955425-947541 From: "442074904605" ;tag=as08373072 Call-ID: 5da9f87d1004dd9e31c9e257573f4261@89.202.128.200 CSeq: 102 INVITE Contact: sip:442070605610@83.245.6.83:5060 Allow-Events: telephone-event Via: SIP/2.0/UDP 89.202.128.200:5060;branch=z9hG4bK07afb6d9;rport Content-Length: 0 <-------------> --- (9 headers 0 lines) --- m900a*CLI> -- SIP/gamma-OUT-08286dc8 is ringing m900a*CLI> -- Local/442070605610@gamma-c7a2,1 is ringing <--- Transmitting (no NAT) to 83.245.6.83:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 83.245.6.83:5060;branch=z9hG4bKff453fd2d4ffb4804d02ab3e894c16a3;received=83.245.6.83 From: ;tag=3373955424-319415 To: ;tag=as7a6f76f4 Call-ID: 1518484-3373955424-319378@msw8.mydomain.com CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY m900a*CLI> Supported: replaces Contact: Content-Length: 0 <------------> m900a*CLI> <--- SIP read from 83.245.6.83:5060 ---> SIP/2.0 200 OK To: ;tag=3373955425-947541 From: "442074904605" ;tag=as08373072 Call-ID: 5da9f87d1004dd9e31c9e257573f4261@89.202.128.200 CSeq: 102 INVITE Contact: sip:442070605610@83.245.6.83:5060 Allow-Events: telephone-event Content-Type: application/sdp Via: SIP/2.0/UDP 89.202.128.200:5060;branch=z9hG4bK07afb6d9;rport Content-Length: 197 v=0 o=msw 9000 7562 IN IP4 83.245.6.83 s=sip call c=IN IP4 83.245.6.84 t=0 0 m=audio 62300 RTP/AVP 0 101 a=ptime:20 a=fmtp:101 0-16 a=rtpmap:101 telephone-event/8000 a=rtpmap:0 PCMU/8000 <-------------> --- (10 headers 10 lines) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 83.245.6.84:62300 Got unsupported a:fmtp in SDP offer Found description format telephone-event for ID 101 Found description format PCMU for ID 0 Capabilities: us - 0x3f1fff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|g722|jpeg|png|h261|h263|h263p|h264), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 83.245.6.84:62300 --- set_address_from_contact host '83.245.6.83' list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 83.245.6.83, port 5060 Transmitting (no NAT) to 83.245.6.83:5060: ACK sip:442070605610@83.245.6.83:5060 SIP/2.0 Via: SIP/2.0/UDP 89.202.128.200:5060;branch=z9hG4bK16462bde;rport Max-Forwards: 70 From: "442074904605" ;tag=as08373072 To: ;tag=3373955425-947541 Contact: Call-ID: 5da9f87d1004dd9e31c9e257573f4261@89.202.128.200 CSeq: 102 ACK User-Agent: Asterisk PBX Remote-Party-ID: "442074904605" ;privacy=off;screen=yes Content-Length: 0 --- m900a*CLI> -- SIP/gamma-OUT-08286dc8 answered Local/442070605610@gamma-c7a2,2 m900a*CLI> -- Local/442070605610@gamma-c7a2,1 answered SIP/83.245.6.83-0828a348 Audio is at 89.202.128.200 port 10526 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 83.245.6.83:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 83.245.6.83:5060;branch=z9hG4bKff453fd2d4ffb4804d02ab3e894c16a3;received=83.245.6.83 From: ;tag=3373955424-319415 To: ;tag=as7a6f76f4 Call-ID: 1518484-3373955424-319378@msw8.mydomain.com CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 244 v=0 o=root 20874 20874 IN IP4 89.202.128.200 s=session c=IN IP4 89.202.128.200 t=0 0 m=audio 10526 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> m900a*CLI> <--- SIP read from 83.245.6.83:5060 ---> ACK sip:448001180118@89.202.128.200 SIP/2.0 Max-Forwards: 11 To: ;tag=as7a6f76f4 From: ;tag=3373955424-319415 Call-ID: 1518484-3373955424-319378@msw8.mydomain.com CSeq: 1 ACK Via: SIP/2.0/UDP 83.245.6.83:5060;branch=z9hG4bK79de99721f1b6d7b9e3ea15dc8928eb7 Contact: sip:442074904605@83.245.6.83:5060 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- m900a*CLI> <--- SIP read from 83.245.6.83:5060 ---> INVITE sip:442074904605@89.202.128.200 SIP/2.0 Max-Forwards: 69 Session-Expires: 3600;Refresher=uac Supported: timer To: "442074904605" ;tag=as08373072 From: ;tag=3373955425-947541 Call-ID: 5da9f87d1004dd9e31c9e257573f4261@89.202.128.200 CSeq: 2 INVITE Via: SIP/2.0/UDP 83.245.6.83:5060;branch=z9hG4bKad39260301605f51f88a54a024d668bc Contact: sip:442070605610@83.245.6.83:5060 Allow-Events: telephone-event Content-Type: application/sdp Content-Length: 346 v=0 o=msw 9000 7563 IN IP4 83.245.6.83 s=sip call c=IN IP4 83.245.6.84 t=0 0 m=image 62300 udptl t38 a=T38FaxUdpEC:t38UDPRedundancy a=T38FaxMaxDatagram:72 a=T38FaxMaxBuffer:200 a=T38FaxRateManagement:transferredTCF a=T38FaxTranscodingJBIG:0 a=T38FaxTranscodingMMR:0 a=T38FaxFillBitRemoval:0 a=T38MaxBitRate:14400 a=T38FaxVersion:0 <-------------> --- (13 headers 15 lines) --- Sending to 83.245.6.83 : 5060 (no NAT) Got T.38 offer in SDP in dialog 5da9f87d1004dd9e31c9e257573f4261@89.202.128.200 Got T.38 Re-invite without audio. Keeping RTP active during T.38 session. Callid 5da9f87d1004dd9e31c9e257573f4261@89.202.128.200 Capabilities: us - 0x3f1fff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|g722|jpeg|png|h261|h263|h263p|h264), peer - audio=0x0 (nothing)/video=0x0 (nothing), combined - 0x0 (nothing) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) <--- Reliably Transmitting (no NAT) to 83.245.6.83:5060 ---> SIP/2.0 488 Not acceptable here Via: SIP/2.0/UDP 83.245.6.83:5060;branch=z9hG4bKad39260301605f51f88a54a024d668bc;received=83.245.6.83 From: ;tag=3373955425-947541 To: "442074904605" ;tag=as08373072 Call-ID: 5da9f87d1004dd9e31c9e257573f4261@89.202.128.200 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 <------------> m900a*CLI> <--- SIP read from 83.245.6.83:5060 ---> ACK sip:442074904605@89.202.128.200 SIP/2.0 Max-Forwards: 69 From: ;tag=3373955425-947541 To: "442074904605" ;tag=as08373072 Call-ID: 5da9f87d1004dd9e31c9e257573f4261@89.202.128.200 CSeq: 2 ACK Via: SIP/2.0/UDP 83.245.6.83:5060;branch=z9hG4bKad39260301605f51f88a54a024d668bc Content-Length: 0 <-------------> --- (8 headers 0 lines) --- m900a*CLI> <--- SIP read from 208.51.25.114:5060 ---> OPTIONS sip:89.202.128.200:5060 SIP/2.0 Via: SIP/2.0/UDP 208.51.25.114:5060;branch=z9hG4bK0go4jn20a0pgihgj4241.1 Call-ID: 6e8f7eb39ab9b7d471747d5dd4a76662@208.51.25.114 To: sip:ping@89.202.128.200 From: sip:ping@208.51.25.114;tag=108352804eb67c3de2e195c22418591f Max-Forwards: 70 CSeq: 12874 OPTIONS <-------------> --- (7 headers 0 lines) --- Looking for s in default (domain 89.202.128.200) <--- Transmitting (no NAT) to 208.51.25.114:5060 ---> SIP/2.0 200 OK Via