Sip read: I> INVITE sip:2003@172.16.254.96;user=phone;phone-context=unknown SIP/2.0 Via: SIP/2.0/UDP 200.61.32.142:5060 From: "52880472" To: Date: Wed, 22 Oct 2003 09:19:49 ARBUE Call-ID: DDEA1FF7-3C011D8-BEBEADBB-C9B38AEA@200.61.32.142 Cisco-Guid: 3723068367-62919128-3200036283-3383986922 User-Agent: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled CSeq: 101 INVITE Max-Forwards: 6 Timestamp: 1066825189 Contact: Expires: 180 Content-Type: application/sdp Content-Length: 166 v=0 o=CiscoSystemsSIP-GW-UserAgent 624 4121 IN IP4 200.61.32.142 s=SIP Call c=IN IP4 200.61.32.142 t=0 0 m=audio 20476 RTP/AVP 8 0 18 65535 65535 65535 4 65535 15 headers, 6 lines Using latest request as basis request Sending to 200.61.32.142 : 5060 (non-NAT) Found audio format ALAW Found audio format UNKN Found audio format UNKN Found audio format UNKN Found audio format UNKN Found audio format UNKN Found audio format ULAW Found audio format UNKN Capabilities: us - 524302, them - 269/854015, combined - 524302 Non-codec capabilities: us - 1, them - 0, combined - 1 Looking for 2003 in default list_route: hop: Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 200.61.32.142:5060 From: "52880472" To: ;tag=as12560d82 Call-ID: DDEA1FF7-3C011D8-BEBEADBB-C9B38AEA@200.61.32.142 CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 200.61.32.142:5060 -- Executing Dial("SIP/-081221b0", "SIP/99952880474@200.61.32.142") in new stack We're at 172.16.254.96 port 15740 Video is at 172.16.254.96 port 12476 Answering with capability 2 Answering with capability 4 Answering with capability 8 Answering with non-codec capability 1 11 headers, 12 lines Reliably Transmitting: INVITE sip:99952880474@200.61.32.142 SIP/2.0 Via: SIP/2.0/UDP 172.16.254.96:5060;branch=z9hG4bK67299947 From: "52880472" ;tag=as7493951b To: Contact: Call-ID: 72a27d622d6836df2d7a7c2c21b39b14@172.16.254.96 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 261 v=0 o=root 20500 20500 IN IP4 172.16.254.96 s=session c=IN IP4 172.16.254.96 t=0 0 m=audio 15740 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 m=video 12476 RTP/AVP (no NAT) to 200.61.32.142:5060 -- Called 99952880474@200.61.32.142 Sip read: I> SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.16.254.96:5060;branch=z9hG4bK67299947 From: "52880472" ;tag=as7493951b To: Date: Wed, 22 Oct 2003 09:19:49 ARBUE Call-ID: 72a27d622d6836df2d7a7c2c21b39b14@172.16.254.96 Server: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled CSeq: 102 INVITE Content-Length: 0 9 headers, 0 lines Sip read: I> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 172.16.254.96:5060;branch=z9hG4bK67299947 From: "52880472" ;tag=as7493951b To: Date: Wed, 22 Oct 2003 09:19:49 ARBUE Call-ID: 72a27d622d6836df2d7a7c2c21b39b14@172.16.254.96 Server: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled CSeq: 102 INVITE Content-Length: 0 9 headers, 0 lines -- SIP/200.61.32.142-e38c is ringing Transmitting (no NAT): SIP/2.0 180 Ringing Via: SIP/2.0/UDP 200.61.32.142:5060 From: "52880472" To: ;tag=as12560d82 Call-ID: DDEA1FF7-3C011D8-BEBEADBB-C9B38AEA@200.61.32.142 CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 200.61.32.142:5060 We're at 172.16.254.96 port 15664 Video is at 172.16.254.96 port 19090 Answering with capability 2 Answering with capability 4 Answering with capability 8 Answering with non-codec capability 1 Transmitting (no NAT): SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 200.61.32.142:5060 From: "52880472" To: ;tag=as12560d82 Call-ID: DDEA1FF7-3C011D8-BEBEADBB-C9B38AEA@200.61.32.142 CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 261 v=0 o=root 20500 20500 IN IP4 172.16.254.96 s=session c=IN IP4 172.16.254.96 t=0 0 m=audio 15664 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 m=video 19090 RTP/AVP to 200.61.32.142:5060 Sip read: I> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.254.96:5060;branch=z9hG4bK67299947 From: "52880472" ;tag=as7493951b To: ;tag=AF869030-7 Date: Wed, 22 Oct 2003 09:19:49 ARBUE Call-ID: 72a27d622d6836df2d7a7c2c21b39b14@172.16.254.96 Server: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled Contact: CSeq: 102 INVITE Content-Type: application/sdp Content-Length: 136 v=0 o=CiscoSystemsSIP-GW-UserAgent 7307 5758 IN IP4 200.61.32.142 s=SIP Call c=IN IP4 200.61.32.142 t=0 0 m=audio 20130 RTP/AVP 0 11 headers, 6 lines Found audio format UNKN Capabilities: us - 524302, them - 4/854015, combined - 524302 Non-codec capabilities: us - 1, them - 0, combined - 1 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 200.61.32.142, port 5060 Transmitting: ACK sip:99952880474@200.61.32.142:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.254.96:5060;branch=z9hG4bK67299947 From: "52880472" ;tag=as7493951b To: ;tag=AF869030-7 Contact: Call-ID: 72a27d622d6836df2d7a7c2c21b39b14@172.16.254.96 CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 200.61.32.142:5060 -- SIP/200.61.32.142-e38c answered SIP/-081221b0 We're at 172.16.254.96 port 15664 Video is at 172.16.254.96 port 19090 Answering with capability 2 Answering with capability 4 Answering with capability 8 Answering with non-codec capability 1 Reliably Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 200.61.32.142:5060 From: "52880472" To: ;tag=as12560d82 Call-ID: DDEA1FF7-3C011D8-BEBEADBB-C9B38AEA@200.61.32.142 CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 261 v=0 o=root 20500 20500 IN IP4 172.16.254.96 s=session c=IN IP4 172.16.254.96 t=0 0 m=audio 15664 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 m=video 19090 RTP/AVP to 200.61.32.142:5060 -- Attempting native bridge of SIP/-081221b0 and SIP/200.61.32.142-e38c Sip read: I> ACK sip:2003@172.16.254.96:5060 SIP/2.0 Via: SIP/2.0/UDP 200.61.32.142:5060 From: "52880472" To: ;tag=as12560d82 Date: Wed, 22 Oct 2003 09:19:49 ARBUE Call-ID: DDEA1FF7-3C011D8-BEBEADBB-C9B38AEA@200.61.32.142 Max-Forwards: 6 Content-Type: application/sdp Content-Length: 135 CSeq: 101 ACK v=0 o=CiscoSystemsSIP-GW-UserAgent 624 4121 IN IP4 200.61.32.142 s=SIP Call c=IN IP4 200.61.32.142 t=0 0 m=audio 20476 RTP/AVP 0 10 headers, 6 lines Found audio format UNKN Capabilities: us - 524302, them - 4/854015, combined - 524302 Non-codec capabilities: us - 1, them - 0, combined - 1 set_destination: Parsing for address/port to send to set_destination: set destination to 200.61.32.142, port 5060 We're at 172.16.254.96 port 15664 Video is at 172.16.254.96 port 19090 Answering with capability 2 Answering with capability 4 Answering with capability 8 Answering with non-codec capability 1 11 headers, 12 lines Reliably Transmitting: INVITE sip:52880472@200.61.32.142:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.254.96:5060;branch=z9hG4bK5b99e531 From: ;tag=as12560d82 To: "52880472" Contact: Call-ID: DDEA1FF7-3C011D8-BEBEADBB-C9B38AEA@200.61.32.142 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 257 v=0 o=root 20500 20500 IN IP4 200.61.32.142 s=session c=IN IP4 200.61.32.142 t=0 0 m=audio 20130 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 m=video 0 RTP/AVP (no NAT) to 200.61.32.142:5060 set_destination: Parsing for address/port to send to set_destination: set destination to 200.61.32.142, port 5060 We're at 172.16.254.96 port 15740 Video is at 172.16.254.96 port 12476 Answering with capability 2 Answering with capability 4 Answering with capability 8 Answering with non-codec capability 1 11 headers, 12 lines Reliably Transmitting: INVITE sip:99952880474@200.61.32.142:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.254.96:5060;branch=z9hG4bK67299947 From: "52880472" ;tag=as7493951b To: ;tag=AF869030-7 Contact: Call-ID: 72a27d622d6836df2d7a7c2c21b39b14@172.16.254.96 CSeq: 103 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 257 v=0 o=root 20500 20500 IN IP4 200.61.32.142 s=session c=IN IP4 200.61.32.142 t=0 0 m=audio 20476 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 m=video 0 RTP/AVP (no NAT) to 200.61.32.142:5060 set_destination: Parsing for address/port to send to set_destination: set destination to 200.61.32.142, port 5060 We're at 172.16.254.96 port 15664 Video is at 172.16.254.96 port 19090 Answering with capability 2 Answering with capability 4 Answering with capability 8 Answering with non-codec capability 1 11 headers, 12 lines Reliably Transmitting: INVITE sip:52880472@200.61.32.142:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.254.96:5060;branch=z9hG4bK5b99e531 From: ;tag=as12560d82 To: "52880472" Contact: Call-ID: DDEA1FF7-3C011D8-BEBEADBB-C9B38AEA@200.61.32.142 CSeq: 103 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 261 v=0 o=root 20500 20500 IN IP4 200.61.32.142 s=session c=IN IP4 200.61.32.142 t=0 0 m=audio 20130 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 m=video 19090 RTP/AVP (no NAT) to 200.61.32.142:5060 set_destination: Parsing for address/port to send to set_destination: set destination to 200.61.32.142, port 5060 We're at 172.16.254.96 port 15740 Video is at 172.16.254.96 port 12476 Answering with capability 2 Answering with capability 4 Answering with capability 8 Answering with non-codec capability 1 11 headers, 12 lines Reliably Transmitting: INVITE sip:99952880474@200.61.32.142:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.254.96:5060;branch=z9hG4bK67299947 From: "52880472" ;tag=as7493951b To: ;tag=AF869030-7 Contact: Call-ID: 72a27d622d6836df2d7a7c2c21b39b14@172.16.254.96 CSeq: 104 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 261 v=0 o=root 20500 20500 IN IP4 200.61.32.142 s=session c=IN IP4 200.61.32.142 t=0 0 m=audio 20476 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 m=video 12476 RTP/AVP (no NAT) to 200.61.32.142:5060 Sip read: I> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.254.96:5060;branch=z9hG4bK5b99e531 From: ;tag=as12560d82 To: "52880472" Date: Wed, 22 Oct 2003 09:19:52 ARBUE Call-ID: DDEA1FF7-3C011D8-BEBEADBB-C9B38AEA@200.61.32.142 Server: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled Contact: CSeq: 102 INVITE Content-Type: application/sdp Content-Length: 135 v=0 o=CiscoSystemsSIP-GW-UserAgent 624 4122 IN IP4 200.61.32.142 s=SIP Call c=IN IP4 200.61.32.142 t=0 0 m=audio 20420 RTP/AVP 0 11 headers, 6 lines Found audio format UNKN Capabilities: us - 524302, them - 4/854015, combined - 524302 Non-codec capabilities: us - 1, them - 0, combined - 1 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 200.61.32.142, port 5060 Transmitting: ACK sip:52880472@200.61.32.142:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.254.96:5060;branch=z9hG4bK5b99e531 From: ;tag=as12560d82 To: "52880472" Contact: Call-ID: DDEA1FF7-3C011D8-BEBEADBB-C9B38AEA@200.61.32.142 CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 200.61.32.142:5060 set_destination: Parsing for address/port to send to set_destination: set destination to 200.61.32.142, port 5060 We're at 172.16.254.96 port 15740 Video is at 172.16.254.96 port 12476 Answering with capability 2 Answering with capability 4 Answering with capability 8 Answering with non-codec capability 1 11 headers, 12 lines Reliably Transmitting: INVITE sip:99952880474@200.61.32.142:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.254.96:5060;branch=z9hG4bK67299947 From: "52880472" ;tag=as7493951b To: ;tag=AF869030-7 Contact: Call-ID: 72a27d622d6836df2d7a7c2c21b39b14@172.16.254.96 CSeq: 105 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 261 v=0 o=root 20500 20500 IN IP4 200.61.32.142 s=session c=IN IP4 200.61.32.142 t=0 0 m=audio 20420 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 m=video 12476 RTP/AVP (no NAT) to 200.61.32.142:5060 Sip read: I> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.254.96:5060;branch=z9hG4bK67299947 From: "52880472" ;tag=as7493951b To: ;tag=AF869030-7 Date: Wed, 22 Oct 2003 09:19:52 ARBUE Call-ID: 72a27d622d6836df2d7a7c2c21b39b14@172.16.254.96 Server: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled Contact: CSeq: 103 INVITE Content-Type: application/sdp Content-Length: 136 v=0 o=CiscoSystemsSIP-GW-UserAgent 7307 5759 IN IP4 200.61.32.142 s=SIP Call c=IN IP4 200.61.32.142 t=0 0 m=audio 20694 RTP/AVP 0 11 headers, 6 lines Found audio format UNKN Capabilities: us - 524302, them - 4/854015, combined - 524302 Non-codec capabilities: us - 1, them - 0, combined - 1 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 200.61.32.142, port 5060 Transmitting: ACK sip:99952880474@200.61.32.142:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.254.96:5060;branch=z9hG4bK67299947 From: "52880472" ;tag=as7493951b To: ;tag=AF869030-7 Contact: Call-ID: 72a27d622d6836df2d7a7c2c21b39b14@172.16.254.96 CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 200.61.32.142:5060 set_destination: Parsing for address/port to send to set_destination: set destination to 200.61.32.142, port 5060 We're at 172.16.254.96 port 15664 Video is at 172.16.254.96 port 19090 Answering with capability 2 Answering with capability 4 Sip read: I> SIP/2.0 481 Invalid CSeq Number Via: SIP/2.0/UDP 172.16.254.96:5060;branch=z9hG4bK5b99e531 From: ;tag=as12560d82 To: "52880472" Call-ID: DDEA1FF7-3C011D8-BEBEADBB-C9B38AEA@200.61.32.142 CSeq: 103 INVITE 6 headers, 0 lines -- Got SIP response 481 "Invalid CSeq Number" back from 200.61.32.142 set_destination: Parsing for address/port to send to set_destination: set destination to 200.61.32.142, port 5060 Transmitting: ACK sip:52880472@200.61.32.142:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.254.96:5060;branch=z9hG4bK5b99e531 From: ;tag=as12560d82 To: "52880472" Contact: Call-ID: DDEA1FF7-3C011D8-BEBEADBB-C9B38AEA@200.61.32.142 CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 200.61.32.142:5060 Answering with capability 8 Answering with non-codec capability 1 11 headers, 12 lines Reliably Transmitting: INVITE sip:52880472@200.61.32.142:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.254.96:5060;branch=z9hG4bK5b99e531 From: ;tag=as12560d82 To: "52880472" Contact: Call-ID: DDEA1FF7-3C011D8-BEBEADBB-C9B38AEA@200.61.32.142 CSeq: 104 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 261 v=0 o=root 20500 20500 IN IP4 200.61.32.142 s=session c=IN IP4 200.61.32.142 t=0 0 m=audio 20694 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 m=video 19090 RTP/AVP (no NAT) to 200.61.32.142:5060 set_destination: Parsing for address/port to send to set_destination: set destination to 200.61.32.142, port 5060 We're at 172.16.254.96 port 15740 Video is at 172.16.254.96 port 12476 Answering with capability 2 Answering with capability 4 Answering with capability 8 Answering with non-codec capability 1 11 headers, 12 lines Reliably Transmitting: INVITE sip:99952880474@200.61.32.142:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.254.96:5060;branch=z9hG4bK67299947 From: "52880472" ;tag=as7493951b To: ;tag=AF869030-7 Contact: Call-ID: 72a27d622d6836df2d7a7c2c21b39b14@172.16.254.96 CSeq: 106 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 261 v=0 o=root 20500 20500 IN IP4 172.16.254.96 s=session c=IN IP4 172.16.254.96 t=0 0 m=audio 15740 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 m=video 12476 RTP/AVP (no NAT) to 200.61.32.142:5060 == Spawn extension (default, 2003, 1) exited non-zero on 'SIP/-081221b0' Sip read: I> SIP/2.0 481 Invalid CSeq Number Via: SIP/2.0/UDP 172.16.254.96:5060;branch=z9hG4bK67299947 From: "52880472" ;tag=as7493951b To: ;tag=AF869030-7 Call-ID: 72a27d622d6836df2d7a7c2c21b39b14@172.16.254.96 CSeq: 104 INVITE 6 headers, 0 lines -- Got SIP response 481 "Invalid CSeq Number" back from 200.61.32.142 set_destination: Parsing for address/port to send to set_destination: set destination to 200.61.32.142, port 5060 Transmitting: ACK sip:99952880474@200.61.32.142:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.254.96:5060;branch=z9hG4bK67299947 From: "52880472" ;tag=as7493951b To: ;tag=AF869030-7 Contact: Call-ID: 72a27d622d6836df2d7a7c2c21b39b14@172.16.254.96 CSeq: 104 ACK User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 200.61.32.142:5060 Sip read: I> SIP/2.0 481 Invalid CSeq Number Via: SIP/2.0/UDP 172.16.254.96:5060;branch=z9hG4bK67299947 From: "52880472" ;tag=as7493951b To: ;tag=AF869030-7 Call-ID: 72a27d622d6836df2d7a7c2c21b39b14@172.16.254.96 CSeq: 105 INVITE 6 headers, 0 lines -- Got SIP response 481 "Invalid CSeq Number" back from 200.61.32.142 set_destination: Parsing for address/port to send to set_destination: set destination to 200.61.32.142, port 5060 Transmitting: ACK sip:99952880474@200.61.32.142:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.254.96:5060;branch=z9hG4bK67299947 From: "52880472" ;tag=as7493951b To: ;tag=AF869030-7 Contact: Call-ID: 72a27d622d6836df2d7a7c2c21b39b14@172.16.254.96 CSeq: 105 ACK User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 200.61.32.142:5060 Sip read: I> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.254.96:5060;branch=z9hG4bK5b99e531 From: ;tag=as12560d82 To: "52880472" Date: Wed, 22 Oct 2003 09:19:52 ARBUE Call-ID: DDEA1FF7-3C011D8-BEBEADBB-C9B38AEA@200.61.32.142 Server: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled Contact: CSeq: 104 INVITE Content-Type: application/sdp Content-Length: 135 v=0 o=CiscoSystemsSIP-GW-UserAgent 624 4123 IN IP4 200.61.32.142 s=SIP Call c=IN IP4 200.61.32.142 t=0 0 m=audio 20978 RTP/AVP 0 11 headers, 6 lines Found audio format UNKN Capabilities: us - 524302, them - 4/854015, combined - 524302 Non-codec capabilities: us - 1, them - 0, combined - 1 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 200.61.32.142, port 5060 Transmitting: ACK sip:52880472@200.61.32.142:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.254.96:5060;branch=z9hG4bK5b99e531 From: ;tag=as12560d82 To: "52880472" Contact: Call-ID: DDEA1FF7-3C011D8-BEBEADBB-C9B38AEA@200.61.32.142 CSeq: 104 ACK User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 200.61.32.142:5060 Sip read: I> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.254.96:5060;branch=z9hG4bK67299947 From: "52880472" ;tag=as7493951b To: ;tag=AF869030-7 Date: Wed, 22 Oct 2003 09:19:52 ARBUE Call-ID: 72a27d622d6836df2d7a7c2c21b39b14@172.16.254.96 Server: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled Contact: CSeq: 106 INVITE Content-Type: application/sdp Content-Length: 136 v=0 o=CiscoSystemsSIP-GW-UserAgent 7307 5760 IN IP4 200.61.32.142 s=SIP Call c=IN IP4 200.61.32.142 t=0 0 m=audio 20826 RTP/AVP 0 11 headers, 6 lines Found audio format UNKN Capabilities: us - 524302, them - 4/854015, combined - 524302 Non-codec capabilities: us - 1, them - 0, combined - 1 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 200.61.32.142, port 5060 Transmitting: ACK sip:99952880474@200.61.32.142:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.254.96:5060;branch=z9hG4bK67299947 From: "52880472" ;tag=as7493951b To: ;tag=AF869030-7 Contact: Call-ID: 72a27d622d6836df2d7a7c2c21b39b14@172.16.254.96 CSeq: 106 ACK User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 200.61.32.142:5060 set_destination: Parsing for address/port to send to set_destination: set destination to 200.61.32.142, port 5060 Reliably Transmitting: BYE sip:99952880474@200.61.32.142:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.254.96:5060;branch=z9hG4bK67299947 From: "52880472" ;tag=as7493951b To: ;tag=AF869030-7 Contact: Call-ID: 72a27d622d6836df2d7a7c2c21b39b14@172.16.254.96 CSeq: 107 BYE User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 200.61.32.142:5060 Sip read: I> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.254.96:5060;branch=z9hG4bK67299947 From: "52880472" ;tag=as7493951b To: ;tag=AF869030-7 Date: Wed, 22 Oct 2003 09:19:52 ARBUE Call-ID: 72a27d622d6836df2d7a7c2c21b39b14@172.16.254.96 Server: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled Content-Length: 0 CSeq: 107 BYE 9 headers, 0 lines Sip read: I> BYE sip:2003@172.16.254.96:5060 SIP/2.0 Via: SIP/2.0/UDP 200.61.32.142:5060 From: "52880472" To: ;tag=as12560d82 Date: Wed, 22 Oct 2003 09:19:52 ARBUE Call-ID: DDEA1FF7-3C011D8-BEBEADBB-C9B38AEA@200.61.32.142 User-Agent: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled Max-Forwards: 6 Timestamp: 1066825212 CSeq: 102 BYE Content-Length: 0 11 headers, 0 lines Sending to 200.61.32.142 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 200.61.32.142:5060 From: "52880472" To: ;tag=as12560d82 Call-ID: DDEA1FF7-3C011D8-BEBEADBB-C9B38AEA@200.61.32.142 CSeq: 102 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 200.61.32.142:5060 noc2pbx*CLI>