=~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2006.07.04 20:52:04 =~=~=~=~=~=~=~=~=~=~=~= <-- SIP read from 212.241.50.12:5060: INVITE sip:0135076777@212.241.37.42 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 212.241.50.12;branch=z9hG4bK0bdb.138ff1a6.0 Via: SIP/2.0/UDP 10.166.38.106:5075;rport=5075;branch=z9hG4bK000005160000bdfe0002 Via: SIP/2.0/UDP 212.241.48.70:5060;branch=z9hG4bK-5307-A4D2 Record-Route: From: ;tag=00-01883-000091d6-670d72a93 To: Call-ID: 00-01883-000091d5-169b9e9d3@tpsip01.ipvoice.enertel.nl CSeq: 25550 INVITE Contact: max-forwards: 29 p-asserted-identity: user-agent: Cirpack/v4.40 (gw_sip) Allow: UPDATE, REFER Content-Type: application/sdp Content-Length: 296 v=0 o=cp10 115203913628 115203913629 IN IP4 10.166.38.107 s=SIP Call c=IN IP4 212.241.50.12 t=0 0 m=audio 53676 RTP/AVP 18 8 101 b=AS:64 a=rtpmap:18 G729/8000/1 a=fmtp:18 annexb=no a=rtpmap:8 PCMA/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=nortpproxy:yes --- (17 headers 14 lines)--- Using INVITE request as basis request - 00-01883-000091d5-169b9e9d3@tpsip01.ipvoice.enertel.nl Sending to 212.241.50.12 : 5060 (non-NAT) Found peer 'global-t.nl' Found RTP audio format 18 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 212.241.50.12:53676 Found description format G729 Found description format PCMA Found description format telephone-event Capabilities: us - 0x108 (alaw|g729), peer - audio=0x108 (alaw|g729)/video=0x0 (nothing), combined - 0x108 (alaw|g729) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 0135076777 in from-pstn (domain 212.241.37.42) list_route: hop: list_route: hop: Transmitting (no NAT) to 212.241.50.12:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 212.241.50.12;branch=z9hG4bK0bdb.138ff1a6.0;received=212.241.50.12 Via: SIP/2.0/UDP 10.166.38.106:5075;rport=5075;branch=z9hG4bK000005160000bdfe0002 Via: SIP/2.0/UDP 212.241.48.70:5060;branch=z9hG4bK-5307-A4D2 From: ;tag=00-01883-000091d6-670d72a93 To: Call-ID: 00-01883-000091d5-169b9e9d3@tpsip01.ipvoice.enertel.nl CSeq: 25550 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- INVITE sip:211@192.1.2.189:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.1.2.10:5060;branch=z9hG4bK3aee568e From: "0652478474" ;tag=as64306e06 To: Contact: Call-ID: 654471313a848ac4731f4ea45a2ddd92@192.1.2.10 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 04 Jul 2006 18:52:16 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 257 v=0 o=root 2013 2013 IN IP4 192.1.2.10 s=session c=IN IP4 192.1.2.10 t=0 0 m=audio 11068 RTP/AVP 18 8 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - <-- SIP read from 192.1.2.189:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.1.2.10:5060;branch=z9hG4bK3aee568e From: "0652478474";tag=as64306e06 To: Call-ID: 654471313a848ac4731f4ea45a2ddd92@192.1.2.10 CSeq: 102 INVITE Content-Length: 0 allium*CLI> <-- SIP read from 192.1.2.189:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.1.2.10:5060;branch=z9hG4bK3aee568e From: "0652478474";tag=as64306e06 To: ;tag=c0a80101-1f20ce Call-ID: 654471313a848ac4731f4ea45a2ddd92@192.1.2.10 CSeq: 102 INVITE Allow: ACK, BYE, CANCEL, INVITE, REFER, OPTIONS, INFO, REGISTER, NOTIFY, UPDATE, SUBSCRIBE, PRACK Contact: Content-Length: 0 <-- SIP read from 192.1.2.189:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.1.2.10:5060;branch=z9hG4bK3aee568e From: "0652478474";tag=as64306e06 To: ;tag=c0a80101-1f20ce Call-ID: 654471313a848ac4731f4ea45a2ddd92@192.1.2.10 CSeq: 102 INVITE Allow: ACK, BYE, CANCEL, INVITE, REFER, OPTIONS, INFO, REGISTER, NOTIFY, UPDATE, SUBSCRIBE, PRACK Contact: Content-Type: application/sdp Content-Length: 181 v=0 o=211 2041046 2041046 IN IP4 192.1.2.189 s=- c=IN IP4 192.1.2.189 t=0 0 m=audio 41000 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 192.1.2.189:41000 Found description format G729 Found description format telephone-event Capabilities: us - 0x108 (alaw|g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) 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 192.1.2.189, port 5060 Transmitting (no NAT) to 192.1.2.189:5060: ACK sip:211@192.1.2.189:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.1.2.10:5060;branch=z9hG4bK3ebb5e17 From: "0652478474" ;tag=as64306e06 To: ;tag=c0a80101-1f20ce Contact: Call-ID: 654471313a848ac4731f4ea45a2ddd92@192.1.2.10 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- allium*CLI> -- SIP/211-8f39 answered SIP/0135076770-4157 --- We're at 212.241.37.42 port 13066 Adding codec 0x100 (g729) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 212.241.50.12:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 212.241.50.12;branch=z9hG4bK0bdb.138ff1a6.0;received=212.241.50.12 Via: SIP/2.0/UDP 10.166.38.106:5075;rport=5075;branch=z9hG4bK000005160000bdfe0002 Via: SIP/2.0/UDP 212.241.48.70:5060;branch=z9hG4bK-5307-A4D2 Record-Route: Record-Route: From: ;tag=00-01883-000091d6-670d72a93 To: ;tag=as5056cd18 Call-ID: 00-01883-000091d5-169b9e9d3@tpsip01.ipvoice.enertel.nl CSeq: 25550 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 263 v=0 o=root 2013 2013 IN IP4 212.241.37.42 s=session c=IN IP4 212.241.37.42 t=0 0 m=audio 13066 RTP/AVP 18 8 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Attempting native bridge of SIP/0135076770-4157 and SIP/211-8f39 --- set_destination: Parsing for address/port to send to set_destination: set destination to 192.1.2.189, port 5060 Reliably Transmitting (no NAT) to 192.1.2.189:5060: BYE sip:211@192.1.2.189:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.1.2.10:5060;branch=z9hG4bK37d50bf9 From: "0652478474" ;tag=as64306e06 To: ;tag=c0a80101-1f20ce Contact: Call-ID: 654471313a848ac4731f4ea45a2ddd92@192.1.2.10 CSeq: 103 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 allium*CLI> <-- SIP read from 192.1.2.189:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.1.2.10:5060;branch=z9hG4bK37d50bf9 From: "0652478474";tag=as64306e06 To: ;tag=c0a80101-1f20ce Call-ID: 654471313a848ac4731f4ea45a2ddd92@192.1.2.10 CSeq: 103 BYE Content-Length: 0 --- (7 headers 0 lines)--- allium*CLI> <-- SIP read from 212.241.50.12:5060: ACK sip:0135076777@212.241.37.42:5060 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 212.241.50.12;branch=0 Via: SIP/2.0/UDP 10.166.38.106:5075;rport=5075;branch=z9hG4bK000005160000bdfe0003 Via: SIP/2.0/UDP 212.241.48.70:5060;branch=z9hG4bK-528C-A4D3 Record-Route: From: ;tag=00-01883-000091d6-670d72a93 To: ;tag=as5056cd18 Call-ID: 00-01883-000091d5-169b9e9d3@tpsip01.ipvoice.enertel.nl CSeq: 25550 ACK Contact: max-forwards: 29 Content-Length: 0 --- (13 headers 0 lines)--- set_destination: Parsing for address/port to send to set_destination: set destination to 212.241.50.12, port 5060 Reliably Transmitting (no NAT) to 212.241.50.12:5060: CANCEL :0135076777@212.241.37.42:5060 SIP/2.0 Via: SIP/2.0/UDP 212.241.37.42:5060;branch=z9hG4bK5249e103;rport Route: , From: To: ;tag=00-01883-000091d6-670d72a93 Contact: Call-ID: 00-01883-000091d5-169b9e9d3@tpsip01.ipvoice.enertel.nl CSeq: 101 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- (7 headers 0 lines)--- allium*CLI> <-- SIP read from 212.241.50.12:5060: SIP/2.0 500 I'm terribly sorry, server error occurred (1/SL) Via: SIP/2.0/UDP 212.241.37.42:5060;branch=z9hG4bK5249e103;rport=5060 From: To: ;tag=00-01883-000091d6-670d72a93 Call-ID: 00-01883-000091d5-169b9e9d3@tpsip01.ipvoice.enertel.nl CSeq: 101 CANCEL Server: OpenSer (1.0.1-notls (i386/linux)) Content-Length: 0 --- (8 headers 0 lines)--- -- Incoming call: Got SIP response 500 "I'm terribly sorry, server error occurred (1/SL)" back from 212.241.50.12 Destroying call '00-01883-000091d5-169b9e9d3@tpsip01.ipvoice.enertel.nl' --- (7 headers 0 lines)--- allium*CLI> <-- SIP read from 212.241.50.12:5060: INVITE sip:0135076777@212.241.37.42:5060 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 212.241.50.12;branch=z9hG4bK1bdb.fbb93df1.0 Via: SIP/2.0/UDP 10.166.38.106:5075;rport=5075;branch=z9hG4bK000005160000bdff0002 Via: SIP/2.0/UDP 212.241.48.70:5060;branch=z9hG4bK-381-A4D5 Record-Route: From: ;tag=00-01883-000091d6-670d72a93 To: ;tag=as5056cd18 Call-ID: 00-01883-000091d5-169b9e9d3@tpsip01.ipvoice.enertel.nl CSeq: 25551 INVITE Contact: max-forwards: 29 p-asserted-identity: user-agent: Cirpack/v4.40 (gw_sip) Allow: UPDATE, REFER Content-Type: application/sdp Content-Length: 296 v=0 o=cp10 115203913628 115203913630 IN IP4 10.166.38.107 s=SIP Call c=IN IP4 212.241.50.12 t=0 0 m=audio 53676 RTP/AVP 18 8 101 b=AS:64 a=rtpmap:18 G729/8000/1 a=fmtp:18 annexb=no a=rtpmap:8 PCMA/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=nortpproxy:yes --- (17 headers 14 lines)--- Using INVITE request as basis request - 00-01883-000091d5-169b9e9d3@tpsip01.ipvoice.enertel.nl Sending to 212.241.50.12 : 5060 (non-NAT) Found peer 'global-t.nl' Found RTP audio format 18 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 212.241.50.12:53676 Found description format G729 Found description format PCMA Found description format telephone-event Capabilities: us - 0x108 (alaw|g729), peer - audio=0x108 (alaw|g729)/video=0x0 (nothing), combined - 0x108 (alaw|g729) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 0135076777 in from-pstn (domain 212.241.37.42) list_route: hop: list_route: hop: Transmitting (no NAT) to 212.241.50.12:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 212.241.50.12;branch=z9hG4bK1bdb.fbb93df1.0;received=212.241.50.12 Via: SIP/2.0/UDP 10.166.38.106:5075;rport=5075;branch=z9hG4bK000005160000bdff0002 Via: SIP/2.0/UDP 212.241.48.70:5060;branch=z9hG4bK-381-A4D5 From: ;tag=00-01883-000091d6-670d72a93 To: ;tag=as5056cd18 Call-ID: 00-01883-000091d5-169b9e9d3@tpsip01.ipvoice.enertel.nl CSeq: 25551 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 INVITE sip:211@192.1.2.189:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.1.2.10:5060;branch=z9hG4bK46ba9707 From: "0652478474" ;tag=as7a3ec4ac To: Contact: Call-ID: 1dd40a5c0c05fe547801bcbc3a94f9ad@192.1.2.10 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 04 Jul 2006 18:52:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 257 v=0 o=root 2013 2013 IN IP4 192.1.2.10 s=session c=IN IP4 192.1.2.10 t=0 0 m=audio 11220 RTP/AVP 18 8 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - <-- SIP read from 192.1.2.189:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.1.2.10:5060;branch=z9hG4bK46ba9707 From: "0652478474";tag=as7a3ec4ac To: Call-ID: 1dd40a5c0c05fe547801bcbc3a94f9ad@192.1.2.10 CSeq: 102 INVITE Content-Length: 0 <-- SIP read from 192.1.2.189:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.1.2.10:5060;branch=z9hG4bK46ba9707 From: "0652478474";tag=as7a3ec4ac To: ;tag=c0a80101-1f25c0 Call-ID: 1dd40a5c0c05fe547801bcbc3a94f9ad@192.1.2.10 CSeq: 102 INVITE Allow: ACK, BYE, CANCEL, INVITE, REFER, OPTIONS, INFO, REGISTER, NOTIFY, UPDATE, SUBSCRIBE, PRACK Contact: Content-Length: 0 <-- SIP read from 192.1.2.181:2051: SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.1.2.10:5060;branch=z9hG4bK06441d18 From: ;tag=as557dbbb1 To: ;tag=c7xtd37eq7 Call-ID: 3c267029ecd1-99z15lq3zijw@snom360-0004132353CF CSeq: 150 NOTIFY Content-Length: 0 <-- SIP read from 192.1.2.189:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.1.2.10:5060;branch=z9hG4bK46ba9707 From: "0652478474";tag=as7a3ec4ac To: ;tag=c0a80101-1f25c0 Call-ID: 1dd40a5c0c05fe547801bcbc3a94f9ad@192.1.2.10 CSeq: 102 INVITE Allow: ACK, BYE, CANCEL, INVITE, REFER, OPTIONS, INFO, REGISTER, NOTIFY, UPDATE, SUBSCRIBE, PRACK Contact: Content-Type: application/sdp Content-Length: 181 v=0 o=211 2042767 2042767 IN IP4 192.1.2.189 s=- c=IN IP4 192.1.2.189 t=0 0 m=audio 41002 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv --- (10 headers 9 lines)--- Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 192.1.2.189:41002 Found description format G729 Found description format telephone-event Capabilities: us - 0x108 (alaw|g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) 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 192.1.2.189, port 5060 Transmitting (no NAT) to 192.1.2.189:5060: ACK sip:211@192.1.2.189:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.1.2.10:5060;branch=z9hG4bK050514d6 From: "0652478474" ;tag=as7a3ec4ac To: ;tag=c0a80101-1f25c0 Contact: Call-ID: 1dd40a5c0c05fe547801bcbc3a94f9ad@192.1.2.10 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- allium*CLI> -- SIP/211-4cf3 answered SIP/0135076770-896f Scheduling destruction of call '6e0b25ba1a04438f26d3fa1e4c0d876d@192.1.2.10' in 32000 ms We're at 212.241.37.42 port 13388 Adding codec 0x100 (g729) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 212.241.50.12:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 212.241.50.12;branch=z9hG4bK1bdb.fbb93df1.0;received=212.241.50.12 Via: SIP/2.0/UDP 10.166.38.106:5075;rport=5075;branch=z9hG4bK000005160000bdff0002 Via: SIP/2.0/UDP 212.241.48.70:5060;branch=z9hG4bK-381-A4D5 Record-Route: Record-Route: From: ;tag=00-01883-000091d6-670d72a93 To: ;tag=as5056cd18 Call-ID: 00-01883-000091d5-169b9e9d3@tpsip01.ipvoice.enertel.nl CSeq: 25551 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 263 v=0 o=root 2013 2013 IN IP4 212.241.37.42 s=session c=IN IP4 212.241.37.42 t=0 0 m=audio 13388 RTP/AVP 18 8 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Attempting native bridge of SIP/0135076770-896f and SIP/211-4cf3 <-- SIP read from 212.241.50.12:5060: ACK sip:0135076777@212.241.37.42:5060 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 212.241.50.12;branch=0 Via: SIP/2.0/UDP 10.166.38.106:5075;rport=5075;branch=z9hG4bK000005160000bdff0003 Via: SIP/2.0/UDP 212.241.48.70:5060;branch=z9hG4bK-55A4-A4DC Record-Route: From: ;tag=00-01883-000091d6-670d72a93 To: ;tag=as5056cd18 Call-ID: 00-01883-000091d5-169b9e9d3@tpsip01.ipvoice.enertel.nl CSeq: 25551 ACK Contact: max-forwards: 29 Content-Length: 0 --- allium*CLI> Scheduling destruction of call '1eda-c0a80101-1-1@192.1.2.183' in 15000 ms