== Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Found Asterisk CVS-HEAD-03/18/05-10:47:18, Copyright (C) 1999 - 2005 Digium. Written by Mark Spencer ========================================================================= Connected to Asterisk CVS-HEAD-03/18/05-10:47:18 currently running on astlap (pid = 24671) astlap*CLI> Verbosity was 3 and is now 7 astlap*CLI> set verbose 9 astlap*CLI> Verbosity was 7 and is now 9 astlap*CLI> set debug 9 astlap*CLI> Core debug was 0 and is now 9 astlap*CLI> sip debug astlap*CLI> SIP Debugging Enabled astlap*CLI> <-- SIP read from 217.10.79.9:5060: --- (0 headers 0 lines) Nat keepalive --- astlap*CLI> <-- SIP read from 192.168.254.251:5061: INVITE sip:2006@192.168.254.250 SIP/2.0 Via: SIP/2.0/UDP 192.168.254.251:5061;branch=z9hG4bK-115073d3;rport From: Kai-Uwe Jensen ;tag=92a08c23897412aao0 To: Call-ID: 5660b057-78ed1c06@192.168.254.251 CSeq: 101 INVITE Max-Forwards: 70 Contact: Kai-Uwe Jensen Expires: 240 User-Agent: Sipura/SPA2000-2.0.13(g) Content-Length: 409 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura Content-Type: application/sdp v=0 o=- 276837874 276837874 IN IP4 192.168.254.251 s=- c=IN IP4 192.168.254.251 t=0 0 m=audio 9090 RTP/AVP 0 2 8 18 96 97 98 100 101 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729a/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:100 NSE/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv --- (14 headers 18 lines)--- Using latest request as basis request Sending to 192.168.254.251 : 5061 (non-NAT) astlap*CLI> Reliably Transmitting (no NAT) to 192.168.254.251:5061: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.254.251:5061;branch=z9hG4bK-115073d3 From: Kai-Uwe Jensen ;tag=92a08c23897412aao0 To: ;tag=as0922f0b9 Call-ID: 5660b057-78ed1c06@192.168.254.251 CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Proxy-Authenticate: Digest realm="asterisk", nonce="338b4f32" Content-Length: 0 --- Scheduling destruction of call '5660b057-78ed1c06@192.168.254.251' in 15000 ms Found user '2001' astlap*CLI> <-- SIP read from 192.168.254.251:5061: ACK sip:2006@192.168.254.250 SIP/2.0 Via: SIP/2.0/UDP 192.168.254.251:5061;branch=z9hG4bK-115073d3;rport From: Kai-Uwe Jensen ;tag=92a08c23897412aao0 To: ;tag=as0922f0b9 Call-ID: 5660b057-78ed1c06@192.168.254.251 CSeq: 101 ACK Max-Forwards: 70 Contact: Kai-Uwe Jensen User-Agent: Sipura/SPA2000-2.0.13(g) Content-Length: 0 --- (10 headers 0 lines)--- astlap*CLI> <-- SIP read from 192.168.254.251:5061: INVITE sip:2006@192.168.254.250 SIP/2.0 Via: SIP/2.0/UDP 192.168.254.251:5061;branch=z9hG4bK-13ed88e7;rport From: Kai-Uwe Jensen ;tag=92a08c23897412aao0 To: Call-ID: 5660b057-78ed1c06@192.168.254.251 CSeq: 102 INVITE Max-Forwards: 70 Proxy-Authorization: Digest username="2001",realm="asterisk",nonce="338b4f32",uri="sip:2006@192.168.254.250",algorithm=MD5,response="11b851cce2616f28366132df876e2bd9" Contact: Kai-Uwe Jensen Expires: 240 User-Agent: Sipura/SPA2000-2.0.13(g) Content-Length: 409 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura Content-Type: application/sdp v=0 o=- 276837874 276837874 IN IP4 192.168.254.251 s=- c=IN IP4 192.168.254.251 t=0 0 m=audio 9090 RTP/AVP 0 2 8 18 96 97 98 100 101 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729a/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:100 NSE/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv --- (15 headers 18 lines)--- Using latest request as basis request Sending to 192.168.254.251 : 5061 (non-NAT) Found user '2001' Found RTP audio format 0 Found RTP audio format 2 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 96 Found RTP audio format 97 Found RTP audio format 98 Found RTP audio format 100 Found RTP audio format 101 Peer audio RTP is at port 192.168.254.251:9090 Found description format PCMU Found description format G726-32 Found description format PCMA Found description format G729a Found description format G726-40 Found description format G726-24 Found description format G726-16 Found description format NSE Found description format telephone-event Capabilities: us - 0x1c0106 (gsm|ulaw|g729|h261|h263|h263p), peer - audio=0x51c (ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0x104 (ulaw|g729) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) Looking for 2006 in home list_route: hop: astlap*CLI> Transmitting (no NAT) to 192.168.254.251:5061: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.254.251:5061;branch=z9hG4bK-13ed88e7 From: Kai-Uwe Jensen ;tag=92a08c23897412aao0 To: ;tag=as2aef10cf Call-ID: 5660b057-78ed1c06@192.168.254.251 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 --- astlap*CLI> -- Executing Dial("SIP/2001-3a3f", "SIP/2006") in new stack astlap*CLI> We're at 192.168.254.250 port 9002 astlap*CLI> Video is at 192.168.254.250 port 9000 astlap*CLI> Answering/Requesting with root capability 0x4 (ulaw) astlap*CLI> Answering with preferred capability 0x100 (g729) astlap*CLI> Answering with non-codec capability 0x1 (telephone-event) astlap*CLI> 12 headers, 11 lines astlap*CLI> Reliably Transmitting (no NAT) to 192.168.254.252:5060: INVITE sip:2006@192.168.254.252:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.254.250:5060;branch=z9hG4bK3d0edf71;rport From: "Jensen L1" ;tag=as019fbfcb To: Contact: Call-ID: 0979b67e42058e562326050567e363d0@192.168.254.250 CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Fri, 18 Mar 2005 22:25:06 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 247 v=0 o=root 24671 24671 IN IP4 192.168.254.250 s=session c=IN IP4 192.168.254.250 t=0 0 m=audio 9002 RTP/AVP 0 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- astlap*CLI> -- Called 2006 astlap*CLI> <-- SIP read from 192.168.254.252:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.254.250:5060;branch=z9hG4bK3d0edf71;rport From: "Jensen L1" ;tag=as019fbfcb To: ;tag=6222825C-DDDC9697 CSeq: 102 INVITE Call-ID: 0979b67e42058e562326050567e363d0@192.168.254.250 Contact: User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.4.1 Content-Length: 0 --- (9 headers 0 lines)--- astlap*CLI> <-- SIP read from 192.168.254.252:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.254.250:5060;branch=z9hG4bK3d0edf71;rport From: "Jensen L1" ;tag=as019fbfcb To: ;tag=6222825C-DDDC9697 CSeq: 102 INVITE Call-ID: 0979b67e42058e562326050567e363d0@192.168.254.250 Contact: User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.4.1 Allow-Events: talk,hold,conference Content-Length: 0 --- (10 headers 0 lines)--- astlap*CLI> -- SIP/2006-5019 is ringing Transmitting (no NAT) to 192.168.254.251:5061: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.254.251:5061;branch=z9hG4bK-13ed88e7 From: Kai-Uwe Jensen ;tag=92a08c23897412aao0 To: ;tag=as2aef10cf Call-ID: 5660b057-78ed1c06@192.168.254.251 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 --- astlap*CLI> <-- SIP read from 192.168.254.252:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.254.250:5060;branch=z9hG4bK3d0edf71;rport From: "Jensen L1" ;tag=as019fbfcb To: ;tag=6222825C-DDDC9697 CSeq: 102 INVITE Call-ID: 0979b67e42058e562326050567e363d0@192.168.254.250 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.4.1 Content-Type: application/sdp Content-Length: 193 v=0 o=- 1111184705 1111184705 IN IP4 192.168.254.252 s=Polycom IP Phone c=IN IP4 192.168.254.252 t=0 0 m=audio 2248 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 --- (11 headers 8 lines)--- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 192.168.254.252:2248 Found description format PCMU Found description format telephone-event Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) 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 192.168.254.252, port 5060 Transmitting (no NAT) to 192.168.254.252:5060: ACK sip:2006@192.168.254.252:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.254.250:5060;branch=z9hG4bK0fc01c46;rport From: "Jensen L1" ;tag=as019fbfcb To: ;tag=6222825C-DDDC9697 Contact: Call-ID: 0979b67e42058e562326050567e363d0@192.168.254.250 CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 --- astlap*CLI> -- SIP/2006-5019 answered SIP/2001-3a3f We're at 192.168.254.250 port 9012 Video is at 192.168.254.250 port 9018 Answering with preferred capability 0x4 (ulaw) Answering with preferred capability 0x100 (g729) Answering with preferred capability 0x2 (gsm) Answering with preferred capability 0x40000 (h261) Answering with preferred capability 0x80000 (h263) Answering with preferred capability 0x100000 (h263p) Answering with non-codec capability 0x1 (telephone-event) Reliably Transmitting (no NAT) to 192.168.254.251:5061: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.254.251:5061;branch=z9hG4bK-13ed88e7 From: Kai-Uwe Jensen ;tag=92a08c23897412aao0 To: ;tag=as2aef10cf Call-ID: 5660b057-78ed1c06@192.168.254.251 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 270 v=0 o=root 24671 24671 IN IP4 192.168.254.250 s=session c=IN IP4 192.168.254.250 t=0 0 m=audio 9012 RTP/AVP 0 18 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Attempting native bridge of SIP/2001-3a3f and SIP/2006-5019 astlap*CLI> <-- SIP read from 192.168.254.251:5061: ACK sip:2006@192.168.254.250 SIP/2.0 Via: SIP/2.0/UDP 192.168.254.251:5061;branch=z9hG4bK-d18b9f11;rport From: Kai-Uwe Jensen ;tag=92a08c23897412aao0 To: ;tag=as2aef10cf Call-ID: 5660b057-78ed1c06@192.168.254.251 CSeq: 102 ACK Max-Forwards: 70 Proxy-Authorization: Digest username="2001",realm="asterisk",nonce="338b4f32",uri="sip:2006@192.168.254.250",algorithm=MD5,response="81a1df7cea52dc57cc44742bf5da40d1" Contact: Kai-Uwe Jensen User-Agent: Sipura/SPA2000-2.0.13(g) Content-Length: 0 --- (11 headers 0 lines)--- astlap*CLI> <-- SIP read from 192.168.254.252:5060: BYE sip:2001@192.168.254.250 SIP/2.0 Via: SIP/2.0/UDP 192.168.254.252:5060;branch=z9hG4bK751ff17d9B3449CA From: ;tag=6222825C-DDDC9697 To: "Jensen L1" ;tag=as019fbfcb CSeq: 1 BYE Call-ID: 0979b67e42058e562326050567e363d0@192.168.254.250 Contact: User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.4.1 Max-Forwards: 70 Content-Length: 0 --- (10 headers 0 lines)--- Sending to 192.168.254.252 : 5060 (non-NAT) astlap*CLI> Transmitting (no NAT) to 192.168.254.252:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.254.252:5060;branch=z9hG4bK751ff17d9B3449CA From: ;tag=6222825C-DDDC9697 To: "Jensen L1" ;tag=as019fbfcb Call-ID: 0979b67e42058e562326050567e363d0@192.168.254.250 CSeq: 1 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 --- -- Executing Hangup("SIP/2001-3a3f", "") in new stack Destroying call '0979b67e42058e562326050567e363d0@192.168.254.250' astlap*CLI> set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.254.251, port 5061 Reliably Transmitting (no NAT) to 192.168.254.251:5061: BYE sip:2001@192.168.254.251:5061 SIP/2.0 Via: SIP/2.0/UDP 192.168.254.250:5060;branch=z9hG4bK08cc4623;rport From: ;tag=as2aef10cf To: Kai-Uwe Jensen ;tag=92a08c23897412aao0 Contact: Call-ID: 5660b057-78ed1c06@192.168.254.251 CSeq: 102 BYE User-Agent: Asterisk PBX Content-Length: 0 --- astlap*CLI> <-- SIP read from 192.168.254.251:5061: SIP/2.0 200 OK To: Kai-Uwe Jensen ;tag=92a08c23897412aao0 From: ;tag=as2aef10cf Call-ID: 5660b057-78ed1c06@192.168.254.251 CSeq: 102 BYE Via: SIP/2.0/UDP 192.168.254.250:5060;branch=z9hG4bK08cc4623;rport=5060 Server: Sipura/SPA2000-2.0.13(g) Content-Length: 0 --- (8 headers 0 lines)--- Message is BYE Destroying call '5660b057-78ed1c06@192.168.254.251' astlap*CLI> sip no d <-- SIP read from 217.10.79.9:5060: --- (0 headers 0 lines) Nat keepalive --- astlap*CLI> sip no debug astlap*CLI> SIP Debugging Disabled astlap*CLI> exit Executing last minute cleanups