Asterisk CVS-D2004.12.06.08.49.16-12/21/04-09:55:48, Copyright (C) 1999-2004 Digium. Written by Mark Spencer ========================================================================= Connected to Asterisk CVS-D2004.12.06.08.49.16-12/21/04-09:55:48 currently running on voip (pid = 22293) Verbosity was 5 and is now 20 Asterisk Ready. -- Remote UNIX connection voip*CLI> sip debug SIP Debugging Enabled voip*CLI> Sip read: INVITE sip:1@xxxxxxxxxxxxx.dyndns.org;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.1.102:5060 From: sergio ;tag=1316098420 To: Call-ID: 2869084273@81.174.4.xxx CSeq: 1 INVITE Contact: sergio User-Agent: Cisco-CP7905/1.02-040406A Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER Expires: 300 Content-Length: 274 Content-Type: application/sdp v=0 o=9 24022 24022 IN IP4 192.168.1.102 s=Cisco 7905 SIP Call c=IN IP4 81.174.4.xxx t=0 0 m=audio 18000 RTP/AVP 8 0 18 101 a=rtpmap:8 PCMA/8000/1 a=rtpmap:0 PCMU/8000/1 a=rtpmap:18 G729/8000/1 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 12 headers, 12 lines Using latest request as basis request Sending to 192.168.1.102 : 5060 (NAT) Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 81.174.4.xxx:18000 Found description format PCMA Found description format PCMU Found description format G729 Found description format telephone-event Capabilities: us - 0x40e (gsm|ulaw|alaw|ilbc), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) Reliably Transmitting (NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.1.102:5060;received=81.174.4.xxx;rport=5060 From: sergio ;tag=1316098420 To: ;tag=as0e3143f6 Call-ID: 2869084273@81.174.4.xxx CSeq: 1 INVITE User-Agent: "ChersoVoip" Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Proxy-Authenticate: Digest realm="asterisk", nonce="686c6bd2" Content-Length: 0 to 81.174.4.xxx:5060 Scheduling destruction of call '2869084273@81.174.4.xxx' in 15000 ms Found user '9' voip*CLI> Sip read: ACK sip:1@xxxxxxxxxxxxx.dyndns.org;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.1.102:5060;received=81.174.4.xxx;rport=5060 From: sergio ;tag=1316098420 To: ;tag=as0e3143f6 Call-ID: 2869084273@81.174.4.xxx CSeq: 1 ACK User-Agent: Cisco-CP7905/1.02-040406A Content-Length: 0 8 headers, 0 lines voip*CLI> Sip read: INVITE sip:1@xxxxxxxxxxxxx.dyndns.org;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.1.102:5060 From: sergio ;tag=1316098420 To: Call-ID: 2869084273@81.174.4.xxx CSeq: 2 INVITE Contact: sergio User-Agent: Cisco-CP7905/1.02-040406A Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER Proxy-Authorization: Digest username="9",realm="asterisk",nonce="686c6bd2",uri="sip:1@xxxxxxxxxxxxx.dyndns.org",response="e9cd7f167efb11f946a6a432b9fafbf9" Expires: 300 Content-Length: 274 Content-Type: application/sdp v=0 o=9 24037 24037 IN IP4 192.168.1.102 s=Cisco 7905 SIP Call c=IN IP4 81.174.4.xxx t=0 0 m=audio 18000 RTP/AVP 8 0 18 101 a=rtpmap:8 PCMA/8000/1 a=rtpmap:0 PCMU/8000/1 a=rtpmap:18 G729/8000/1 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 13 headers, 12 lines Using latest request as basis request Sending to 192.168.1.102 : 5060 (NAT) Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 81.174.4.xxx:18000 Found description format PCMA Found description format PCMU Found description format G729 Found description format telephone-event Capabilities: us - 0x40e (gsm|ulaw|alaw|ilbc), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) Found user '9' Looking for 1 in sip list_route: hop: Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.102:5060;received=81.174.4.xxx;rport=5060 From: sergio ;tag=1316098420 To: ;tag=as57747088 Call-ID: 2869084273@81.174.4.xxx CSeq: 2 INVITE User-Agent: "ChersoVoip" Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 81.174.4.xxx:5060 -- Executing NoOp("SIP/9-4a8e", "Chiamata a interni SIP") in new stack -- Executing Macro("SIP/9-4a8e", "interni|SIP|1|90") in new stack -- Executing NoCDR("SIP/9-4a8e", "") in new stack Dec 21 14:49:49 WARNING[22340]: cdr.c:114 ast_cdr_free: CDR on channel 'SIP/9-4a8e' not posted Dec 21 14:49:49 WARNING[22340]: cdr.c:116 ast_cdr_free: CDR on channel 'SIP/9-4a8e' lacks end -- Executing SetVar("SIP/9-4a8e", "_ALERT_INFO=") in new stack -- Executing Answer("SIP/9-4a8e", "") in new stack We're at 82.50.122.xxx port 18336 Answering with preferred capability 0x8 (alaw) Answering with preferred capability 0x4 (ulaw) Answering with preferred capability 0x100 (g729) Answering with non-codec capability 0x1 (telephone-event) Reliably Transmitting (NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.102:5060;received=81.174.4.xxx;rport=5060 From: sergio ;tag=1316098420 To: ;tag=as57747088 Call-ID: 2869084273@81.174.4.xxx CSeq: 2 INVITE User-Agent: "ChersoVoip" Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 266 v=0 o=root 22340 22340 IN IP4 82.50.122.xxx s=session c=IN IP4 82.50.122.xxx t=0 0 m=audio 18336 RTP/AVP 8 0 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - to 81.174.4.xxx:5060 -- Executing Dial("SIP/9-4a8e", "SIP/1|90|Tt") in new stack We're at 192.168.0.100 port 17710 Answering/Requesting with root capability 8 Answering with preferred capability 0x4 (ulaw) Answering with capability 0x100 (g729) Answering with non-codec capability 0x1 (telephone-event) 13 headers, 12 lines Reliably Transmitting: INVITE sip:1@192.168.0.101:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.100:5060;branch=z9hG4bK6f3411bd From: "sergio" ;tag=as3940525f To: Contact: Call-ID: 099bc783479be2e36de32b4f21075b1f@192.168.0.100 CSeq: 102 INVITE User-Agent: "ChersoVoip" Date: Tue, 21 Dec 2004 13:49:49 GMT Alert-info: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 268 =0ip*CLI> o=root 22340 22340 IN IP4 192.168.0.100 s=session c=IN IP4 192.168.0.100 t=0 0 m=audio 17710 RTP/AVP 8 0 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - (no NAT) to 192.168.0.101:5060 -- Called 1 voip*CLI> Sip read: ACK sip:1@82.50.122.xxx SIP/2.0 Via: SIP/2.0/UDP 192.168.1.102:5060 From: sergio ;tag=1316098420 To: ;tag=as57747088 Call-ID: 2869084273@81.174.4.xxx CSeq: 2 ACK User-Agent: Cisco-CP7905/1.02-040406A Proxy-Authorization: Digest username="9",realm="asterisk",nonce="686c6bd2",uri="sip:1@xxxxxxxxxxxxx.dyndns.org",response="e9cd7f167efb11f946a6a432b9fafbf9" Content-Length: 0 9 headers, 0 lines voip*CLI> Sip read: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.100:5060;branch=z9hG4bK6f3411bd From: "sergio" ;tag=as3940525f To: Call-ID: 099bc783479be2e36de32b4f21075b1f@192.168.0.100 Date: Tue, 21 Dec 2004 13:49:38 GMT CSeq: 102 INVITE Server: CSCO/7 Contact: Content-Length: 0 10 headers, 0 lines voip*CLI> Sip read: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.0.100:5060;branch=z9hG4bK6f3411bd From: "sergio" ;tag=as3940525f To: ;tag=001192d98307001862adf81b-70efd6c4 Call-ID: 099bc783479be2e36de32b4f21075b1f@192.168.0.100 Date: Tue, 21 Dec 2004 13:49:38 GMT CSeq: 102 INVITE Server: CSCO/7 Contact: Content-Length: 0 10 headers, 0 lines -- SIP/1-fa01 is ringing voip*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.100:5060;branch=z9hG4bK6f3411bd From: "sergio" ;tag=as3940525f To: ;tag=001192d98307001862adf81b-70efd6c4 Call-ID: 099bc783479be2e36de32b4f21075b1f@192.168.0.100 Date: Tue, 21 Dec 2004 13:49:40 GMT CSeq: 102 INVITE Server: CSCO/7 Contact: Content-Type: application/sdp Content-Length: 198 v=0 o=Cisco-SIPUA 4843 12777 IN IP4 192.168.0.101 s=SIP Call c=IN IP4 192.168.0.101 t=0 0 m=audio 25542 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 11 headers, 9 lines Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 192.168.0.101:25542 Found description format PCMA Found description format telephone-event Capabilities: us - 0x10c (ulaw|alaw|g729), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.0.101, port 5060 Transmitting: ACK sip:1@192.168.0.101:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.100:5060;branch=z9hG4bK3636f53c From: "sergio" ;tag=as3940525f To: ;tag=001192d98307001862adf81b-70efd6c4 Contact: Call-ID: 099bc783479be2e36de32b4f21075b1f@192.168.0.100 Seq: 102 ACK User-Agent: "ChersoVoip" Content-Length: 0 (no NAT) to 192.168.0.101:5060 -- SIP/1-fa01 answered SIP/9-4a8e -- Attempting native bridge of SIP/9-4a8e and SIP/1-fa01 voip*CLI> Sip read: INVITE sip:9@192.168.0.100:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK16eef2ab From: ;tag=001192d98307001862adf81b-70efd6c4 To: "sergio" ;tag=as3940525f Call-ID: 099bc783479be2e36de32b4f21075b1f@192.168.0.100 Date: Tue, 21 Dec 2004 13:49:49 GMT CSeq: 101 INVITE User-Agent: CSCO/7 Contact: Content-Type: application/sdp Content-Length: 193 v=0 o=Cisco-SIPUA 14040 23367 IN IP4 192.168.0.101 s=SIP Call c=IN IP4 0.0.0.0 t=0 0 m=audio 25542 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 11 headers, 9 lines Using latest request as basis request Sending to 192.168.0.101 : 5060 (non-NAT) Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 0.0.0.0:25542 Found description format PCMA Found description format telephone-event Capabilities: us - 0x10c (ulaw|alaw|g729), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) -- Started music on hold, class 'default', on SIP/9-4a8e We're at 192.168.0.100 port 17710 Answering/Requesting with root capability 8 Answering with preferred capability 0x4 (ulaw) Answering with capability 0x100 (g729) Answering with non-codec capability 0x1 (telephone-event) Reliably Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK16eef2ab From: ;tag=001192d98307001862adf81b-70efd6c4 To: "sergio" ;tag=as3940525f Call-ID: 099bc783479be2e36de32b4f21075b1f@192.168.0.100 CSeq: 101 INVITE User-Agent: "ChersoVoip" Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 268 v=0 o=root 22340 22341 IN IP4 192.168.0.100 s=session c=IN IP4 192.168.0.100 t=0 0 m=audio 17710 RTP/AVP 8 0 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - to 192.168.0.101:5060 voip*CLI> Sip read: ACK sip:9@192.168.0.100:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK304e6a6b From: ;tag=001192d98307001862adf81b-70efd6c4 To: "sergio" ;tag=as3940525f Call-ID: 099bc783479be2e36de32b4f21075b1f@192.168.0.100 Date: Tue, 21 Dec 2004 13:49:49 GMT CSeq: 101 ACK User-Agent: CSCO/7 Content-Length: 0 9 headers, 0 lines voip*CLI> Sip read: INVITE sip:9@192.168.0.100:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK323f6eca From: ;tag=001192d98307001862adf81b-70efd6c4 To: "sergio" ;tag=as3940525f Call-ID: 099bc783479be2e36de32b4f21075b1f@192.168.0.100 Date: Tue, 21 Dec 2004 13:49:52 GMT CSeq: 102 INVITE User-Agent: CSCO/7 Contact: Content-Type: application/sdp Content-Length: 249 v=0 o=Cisco-SIPUA 24798 25449 IN IP4 192.168.0.101 s=SIP Call c=IN IP4 192.168.0.101 t=0 0 m=audio 25542 RTP/AVP 8 0 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 11 headers, 11 lines Using latest request as basis request Sending to 192.168.0.101 : 5060 (non-NAT) Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 192.168.0.101:25542 Found description format PCMA Found description format PCMU Found description format G729 Found description format telephone-event Capabilities: us - 0x10c (ulaw|alaw|g729), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x10c (ulaw|alaw|g729) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) -- Stopped music on hold on SIP/9-4a8e We're at 192.168.0.100 port 17710 Answering/Requesting with root capability 8 Answering with preferred capability 0x4 (ulaw) Answering with capability 0x100 (g729) Answering with non-codec capability 0x1 (telephone-event) Reliably Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK323f6eca From: ;tag=001192d98307001862adf81b-70efd6c4 To: "sergio" ;tag=as3940525f Call-ID: 099bc783479be2e36de32b4f21075b1f@192.168.0.100 CSeq: 102 INVITE User-Agent: "ChersoVoip" Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 268 v=0 o=root 22340 22342 IN IP4 192.168.0.100 s=session c=IN IP4 192.168.0.100 t=0 0 m=audio 17710 RTP/AVP 8 0 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - to 192.168.0.101:5060 voip*CLI> Sip read: ACK sip:9@192.168.0.100:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK2709ae0e From: ;tag=001192d98307001862adf81b-70efd6c4 To: "sergio" ;tag=as3940525f Call-ID: 099bc783479be2e36de32b4f21075b1f@192.168.0.100 Date: Tue, 21 Dec 2004 13:49:52 GMT CSeq: 102 ACK User-Agent: CSCO/7 Content-Length: 0 9 headers, 0 lines voip*CLI> Sip read: BYE sip:9@192.168.0.100:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK31c162a6 From: ;tag=001192d98307001862adf81b-70efd6c4 To: "sergio" ;tag=as3940525f Call-ID: 099bc783479be2e36de32b4f21075b1f@192.168.0.100 Date: Tue, 21 Dec 2004 13:49:56 GMT CSeq: 103 BYE User-Agent: CSCO/7 Content-Length: 0 9 headers, 0 lines Sending to 192.168.0.101 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK31c162a6 From: ;tag=001192d98307001862adf81b-70efd6c4 To: "sergio" ;tag=as3940525f Call-ID: 099bc783479be2e36de32b4f21075b1f@192.168.0.100 CSeq: 103 BYE User-Agent: "ChersoVoip" Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 192.168.0.101:5060 == Spawn extension (macro-interni, s, 4) exited non-zero on 'SIP/9-4a8e' in macro 'interni' == Spawn extension (sip, 1, 2) exited non-zero on 'SIP/9-4a8e' set_destination: Parsing for address/port to send to set_destination: set destination to 81.174.4.xxx, port 5060 Reliably Transmitting: BYE sip:9@81.174.4.xxx:5060 SIP/2.0 Via: SIP/2.0/UDP 82.50.122.xxx:5060;branch=z9hG4bK266768c1;rport From: ;tag=as57747088 To: sergio ;tag=1316098420 Contact: Call-ID: 2869084273@81.174.4.xxx CSeq: 102 BYE User-Agent: "ChersoVoip" Content-Length: 0 (NAT) to 81.174.4.xxx:5060 voip*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 82.50.122.xxx:5060;branch=z9hG4bK266768c1;rport From: ;tag=as57747088 To: sergio ;tag=1316098420 Call-ID: 2869084273@81.174.4.xxx CSeq: 102 BYE Server: Cisco-CP7905/1.02-040406A Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER Content-Length: 0 9 headers, 0 lines Message is BYE Destroying call '099bc783479be2e36de32b4f21075b1f@192.168.0.100' Destroying call '2869084273@81.174.4.xxx'