== 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 = 24546) astlap*CLI> Verbosity is at least 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 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-6eaaaca1;rport From: Kai-Uwe Jensen ;tag=27b6fce1d32b02a1o0 To: Call-ID: 325539e1-fc59e9a1@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=- 276828047 276828047 IN IP4 192.168.254.251 s=- c=IN IP4 192.168.254.251 t=0 0 m=audio 9088 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) 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-6eaaaca1 From: Kai-Uwe Jensen ;tag=27b6fce1d32b02a1o0 To: ;tag=as468549ba Call-ID: 325539e1-fc59e9a1@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="04e2aa78" Content-Length: 0 --- Scheduling destruction of call '325539e1-fc59e9a1@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-6eaaaca1;rport From: Kai-Uwe Jensen ;tag=27b6fce1d32b02a1o0 To: ;tag=as468549ba Call-ID: 325539e1-fc59e9a1@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-b582ed89;rport From: Kai-Uwe Jensen ;tag=27b6fce1d32b02a1o0 To: Call-ID: 325539e1-fc59e9a1@192.168.254.251 CSeq: 102 INVITE Max-Forwards: 70 Proxy-Authorization: Digest username="2001",realm="asterisk",nonce="04e2aa78",uri="sip:2006@192.168.254.250",algorithm=MD5,response="403b10dce5be4d4d3e99b84391c1d9cd" 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=- 276828047 276828047 IN IP4 192.168.254.251 s=- c=IN IP4 192.168.254.251 t=0 0 m=audio 9088 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 192.168.254.251:9088 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: 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-b582ed89 From: Kai-Uwe Jensen ;tag=27b6fce1d32b02a1o0 To: ;tag=as3b099092 Call-ID: 325539e1-fc59e9a1@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/2001-49c8", "SIP/2006") in new stack astlap*CLI> We're at 192.168.254.250 port 9044 astlap*CLI> Video is at 192.168.254.250 port 9020 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=z9hG4bK2ca3739b;rport From: "Jensen L1" ;tag=as200ef6d0 To: Contact: Call-ID: 4be840bf558d65927b0ffaa54e247147@192.168.254.250 CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Fri, 18 Mar 2005 22:23:28 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 247 v=0 o=root 24546 24546 IN IP4 192.168.254.250 s=session c=IN IP4 192.168.254.250 t=0 0 m=audio 9044 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=z9hG4bK2ca3739b;rport From: "Jensen L1" ;tag=as200ef6d0 To: ;tag=F9FDAC1B-EC354332 CSeq: 102 INVITE Call-ID: 4be840bf558d65927b0ffaa54e247147@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=z9hG4bK2ca3739b;rport From: "Jensen L1" ;tag=as200ef6d0 To: ;tag=F9FDAC1B-EC354332 CSeq: 102 INVITE Call-ID: 4be840bf558d65927b0ffaa54e247147@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/2006-35c1 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-b582ed89 From: Kai-Uwe Jensen ;tag=27b6fce1d32b02a1o0 To: ;tag=as3b099092 Call-ID: 325539e1-fc59e9a1@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=z9hG4bK2ca3739b;rport From: "Jensen L1" ;tag=as200ef6d0 To: ;tag=F9FDAC1B-EC354332 CSeq: 102 INVITE Call-ID: 4be840bf558d65927b0ffaa54e247147@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=- 1111184609 1111184609 IN IP4 192.168.254.252 s=Polycom IP Phone c=IN IP4 192.168.254.252 t=0 0 m=audio 2246 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 192.168.254.252:2246 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=z9hG4bK01bbee7e;rport From: "Jensen L1" ;tag=as200ef6d0 To: Contact: Call-ID: 4be840bf558d65927b0ffaa54e247147@192.168.254.250 CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 --- astlap*CLI> -- SIP/2006-35c1 answered SIP/2001-49c8 We're at 192.168.254.250 port 9004 Video is at 192.168.254.250 port 9008 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-b582ed89 From: Kai-Uwe Jensen ;tag=27b6fce1d32b02a1o0 To: ;tag=as3b099092 Call-ID: 325539e1-fc59e9a1@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 24546 24546 IN IP4 192.168.254.250 s=session c=IN IP4 192.168.254.250 t=0 0 m=audio 9004 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/2001-49c8 and SIP/2006-35c1 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-6eeb8d55;rport From: Kai-Uwe Jensen ;tag=27b6fce1d32b02a1o0 To: ;tag=as3b099092 Call-ID: 325539e1-fc59e9a1@192.168.254.251 CSeq: 102 ACK Max-Forwards: 70 Proxy-Authorization: Digest username="2001",realm="asterisk",nonce="04e2aa78",uri="sip:2006@192.168.254.250",algorithm=MD5,response="a3f2245784c0c67d85649d2af84c86dd" 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: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.254.250:5060;branch=z9hG4bK2ca3739b;rport From: "Jensen L1" ;tag=as200ef6d0 To: ;tag=F9FDAC1B-EC354332 CSeq: 102 INVITE Call-ID: 4be840bf558d65927b0ffaa54e247147@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=- 1111184609 1111184609 IN IP4 192.168.254.252 s=Polycom IP Phone c=IN IP4 192.168.254.252 t=0 0 m=audio 2246 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 192.168.254.252:2246 astlap*CLI> 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) 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=z9hG4bK0de3890f;rport From: "Jensen L1" ;tag=as200ef6d0 To: Contact: Call-ID: 4be840bf558d65927b0ffaa54e247147@192.168.254.250 CSeq: 102 ACK User-Agent: Asterisk PBX 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=z9hG4bK2ca3739b;rport From: "Jensen L1" ;tag=as200ef6d0 To: ;tag=F9FDAC1B-EC354332 CSeq: 102 INVITE Call-ID: 4be840bf558d65927b0ffaa54e247147@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=- 1111184609 1111184609 IN IP4 192.168.254.252 s=Polycom IP Phone c=IN IP4 192.168.254.252 t=0 0 m=audio 2246 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 192.168.254.252:2246 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) 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=z9hG4bK17e73f60;rport From: "Jensen L1" ;tag=as200ef6d0 To: Contact: Call-ID: 4be840bf558d65927b0ffaa54e247147@192.168.254.250 CSeq: 102 ACK User-Agent: Asterisk PBX 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=z9hG4bK2ca3739b;rport From: "Jensen L1" ;tag=as200ef6d0 To: ;tag=F9FDAC1B-EC354332 CSeq: 102 INVITE Call-ID: 4be840bf558d65927b0ffaa54e247147@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=- 1111184609 1111184609 IN IP4 192.168.254.252 s=Polycom IP Phone c=IN IP4 192.168.254.252 t=0 0 m=audio 2246 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 192.168.254.252:2246 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) set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.254.252, port 5060 astlap*CLI> 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=z9hG4bK22afe6c6;rport From: "Jensen L1" ;tag=as200ef6d0 To: Contact: Call-ID: 4be840bf558d65927b0ffaa54e247147@192.168.254.250 CSeq: 102 ACK User-Agent: Asterisk PBX 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=z9hG4bK2ca3739b;rport From: "Jensen L1" ;tag=as200ef6d0 To: ;tag=F9FDAC1B-EC354332 CSeq: 102 INVITE Call-ID: 4be840bf558d65927b0ffaa54e247147@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=- 1111184609 1111184609 IN IP4 192.168.254.252 s=Polycom IP Phone c=IN IP4 192.168.254.252 t=0 0 m=audio 2246 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 192.168.254.252:2246 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) 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=z9hG4bK5b6e546c;rport From: "Jensen L1" ;tag=as200ef6d0 To: Contact: Call-ID: 4be840bf558d65927b0ffaa54e247147@192.168.254.250 CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 --- astlap*CLI> <-- SIP read from 217.10.79.9:5060: --- (0 headers 0 lines) Nat keepalive --- 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=z9hG4bK2ca3739b;rport From: "Jensen L1" ;tag=as200ef6d0 To: ;tag=F9FDAC1B-EC354332 CSeq: 102 INVITE Call-ID: 4be840bf558d65927b0ffaa54e247147@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=- 1111184609 1111184609 IN IP4 192.168.254.252 s=Polycom IP Phone c=IN IP4 192.168.254.252 t=0 0 m=audio 2246 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 192.168.254.252:2246 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) 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=z9hG4bK461d8e1b;rport From: "Jensen L1" ;tag=as200ef6d0 To: Contact: Call-ID: 4be840bf558d65927b0ffaa54e247147@192.168.254.250 CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 --- astlap*CLI> <-- SIP read from 192.168.254.251:5061: BYE sip:2006@192.168.254.250 SIP/2.0 Via: SIP/2.0/UDP 192.168.254.251:5061;branch=z9hG4bK-44c09495;rport From: Kai-Uwe Jensen ;tag=27b6fce1d32b02a1o0 To: ;tag=as3b099092 Call-ID: 325539e1-fc59e9a1@192.168.254.251 CSeq: 103 BYE Max-Forwards: 70 Proxy-Authorization: Digest username="2001",realm="asterisk",nonce="04e2aa78",uri="sip:2006@192.168.254.250",algorithm=MD5,response="9b64fa976275627403b59e1fb4cf6ea7" User-Agent: Sipura/SPA2000-2.0.13(g) Content-Length: 0 --- (10 headers 0 lines)--- Sending to 192.168.254.251 : 5061 (non-NAT) 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-44c09495 From: Kai-Uwe Jensen ;tag=27b6fce1d32b02a1o0 To: ;tag=as3b099092 Call-ID: 325539e1-fc59e9a1@192.168.254.251 CSeq: 103 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 --- set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.254.252, port 5060 Reliably Transmitting (no NAT) to 192.168.254.252:5060: BYE sip:2006@192.168.254.252:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.254.250:5060;branch=z9hG4bK2b2c4a98;rport From: "Jensen L1" ;tag=as200ef6d0 To: Contact: Call-ID: 4be840bf558d65927b0ffaa54e247147@192.168.254.250 CSeq: 103 BYE User-Agent: Asterisk PBX Content-Length: 0 --- -- Executing Hangup("SIP/2001-49c8", "") in new stack astlap*CLI> <-- SIP read from 192.168.254.252:5060: SIP/2.0 500 Internal Server Error Via: SIP/2.0/UDP 192.168.254.250:5060;branch=z9hG4bK2b2c4a98;rport From: "Jensen L1" ;tag=as200ef6d0 To: CSeq: 103 BYE Call-ID: 4be840bf558d65927b0ffaa54e247147@192.168.254.250 Contact: User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.4.1 Content-Length: 0 --- (9 headers 0 lines)--- -- Got SIP response 500 "Internal Server Error" back from 192.168.254.252 Destroying call '4be840bf558d65927b0ffaa54e247147@192.168.254.250' Destroying call '325539e1-fc59e9a1@192.168.254.251' 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=z9hG4bK2ca3739b;rport From: "Jensen L1" ;tag=as200ef6d0 To: ;tag=F9FDAC1B-EC354332 CSeq: 102 INVITE Call-ID: 4be840bf558d65927b0ffaa54e247147@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=- 1111184609 1111184609 IN IP4 192.168.254.252 s=Polycom IP Phone c=IN IP4 192.168.254.252 t=0 0 m=audio 2246 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 --- (11 headers 8 lines)--- Destroying call '4be840bf558d65927b0ffaa54e247147@192.168.254.250' astlap*CLI> sip no debug astlap*CLI> SIP Debugging Disabled astlap*CLI> exit Executing last minute cleanups