INVITE sip:7001@192.168.0.200;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.0.201:5060 From: ;tag=545885975 To: Call-ID: 1683847546@192.168.0.201 CSeq: 1 INVITE Contact: User-Agent: Cisco ATA 188 v3.1.1 atasip (040629A) Expires: 300 Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER Supported: replaces Content-Length: 249 Content-Type: application/sdp o=2299 11146 11146 IN IP4 192.168.0.201 s=ATA186 Call c=IN IP4 192.168.0.201 t=0 0 m=audio 16384 RTP/AVP 0 4 8 101 a=rtpmap:0 PCMU/8000/1 a=rtpmap:4 G723/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 13 headers, 11 lines Using latest request as basis request Sending to 192.168.0.201 : 5060 (non-NAT) Found RTP audio format 0 Found RTP audio format 4 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 192.168.0.201:16384 Found description format PCMU Found description format G723 Found description format PCMA Found description format telephone-event Capabilities: us - 0x8000e(GSM|ULAW|ALAW|H263), peer - audio=0xd(G723|ULAW|ALAW)/video=0x0(EMPTY), combined - 0xc(ULAW|ALAW) Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723) Found user '2299' Looking for 7001 in default list_route: hop: Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.201:5060 From: ;tag=545885975 To: ;tag=as4728682c Call-ID: 1683847546@192.168.0.201 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 192.168.0.201:5060 We're at 192.168.0.200 port 17084 Video is at 192.168.0.200 port 16692 Answering/Requesting with root capability 8 Answering with capability 0x2(GSM) Answering with capability 0x4(ULAW) Answering with capability 0x80000(H263) Answering with non-codec capability 0x1(G723) 12 headers, 14 lines Reliably Transmitting: INVITE sip:7001@192.168.0.206 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.200:5060;branch=z9hG4bK0ce634f0 From: "ATA Line 1" ;tag=as5539761f To: Contact: Call-ID: 3722563f6b547b605d2b9d6f5f1ad756@192.168.0.200 CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Mon, 16 Aug 2004 20:06:43 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 315 o=root 25260 25260 IN IP4 192.168.0.200 s=session c=IN IP4 192.168.0.200 t=0 0 m=audio 17084 RTP/AVP 8 3 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - m=video 16692 RTP/AVP 34 a=rtpmap:34 H263/90000 (no NAT) to 192.168.0.206:5060 [Kasterisk*CLI> Transmitting (no NAT): SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.0.201:5060 From: ;tag=545885975 To: ;tag=as4728682c Call-ID: 1683847546@192.168.0.201 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 192.168.0.201:5060 [Kasterisk*CLI> Sip read: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.200:5060;branch=z9hG4bK0ce634f0 From: "ATA Line 1" ;tag=as5539761f To: ;tag=464E90-1FE Date: Fri, 01 Mar 2002 01:16:47 GMT Call-ID: 3722563f6b547b605d2b9d6f5f1ad756@192.168.0.200 Server: Cisco-SIPGateway/IOS-12.x CSeq: 102 INVITE Allow-Events: telephone-event Content-Length: 0 10 headers, 0 lines [Kasterisk*CLI> Sip read: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.0.200:5060;branch=z9hG4bK0ce634f0 From: "ATA Line 1" ;tag=as5539761f To: ;tag=464E90-1FE Date: Fri, 01 Mar 2002 01:16:47 GMT Call-ID: 3722563f6b547b605d2b9d6f5f1ad756@192.168.0.200 Server: Cisco-SIPGateway/IOS-12.x CSeq: 102 INVITE Allow: UPDATE Allow-Events: telephone-event Contact: Content-Disposition: session;handling=required Content-Type: application/sdp Content-Length: 281 o=CiscoSystemsSIP-GW-UserAgent 6900 6347 IN IP4 192.168.0.206 s=SIP Call c=IN IP4 192.168.0.206 t=0 0 m=audio 18900 RTP/AVP 8 101 c=IN IP4 192.168.0.206 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 m=video 0 RTP/AVP c=IN IP4 192.168.0.206 14 headers, 12 lines Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 192.168.0.206:18900 Found description format PCMA Found description format telephone-event [Kasterisk*CLI> Capabilities: us - 0x8000e(GSM|ULAW|ALAW|H263), peer - audio=0x8(ALAW)/video=0x0(EMPTY), combined - 0x8(ALAW) Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723) [Kasterisk*CLI> We're at 192.168.0.200 port 12878 Video is at 192.168.0.200 port 18752 Answering with capability 0x8(ALAW) Answering with non-codec capability 0x1(G723) Transmitting (no NAT): SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.0.201:5060 From: ;tag=545885975 To: ;tag=as4728682c Call-ID: 1683847546@192.168.0.201 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 218 o=root 25260 25260 IN IP4 192.168.0.200 s=session c=IN IP4 192.168.0.200 t=0 0 m=audio 12878 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - to 192.168.0.201:5060 [Kasterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.200:5060;branch=z9hG4bK0ce634f0 From: "ATA Line 1" ;tag=as5539761f To: ;tag=464E90-1FE Date: Fri, 01 Mar 2002 01:16:47 GMT Call-ID: 3722563f6b547b605d2b9d6f5f1ad756@192.168.0.200 Server: Cisco-SIPGateway/IOS-12.x CSeq: 102 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER Allow-Events: telephone-event Contact: Content-Type: application/sdp Content-Length: 281 o=CiscoSystemsSIP-GW-UserAgent 6900 6347 IN IP4 192.168.0.206 s=SIP Call c=IN IP4 192.168.0.206 t=0 0 m=audio 18900 RTP/AVP 8 101 c=IN IP4 192.168.0.206 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 m=video 0 RTP/AVP c=IN IP4 192.168.0.206 13 headers, 12 lines Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 192.168.0.206:18900 Found description format PCMA Found description format telephone-event Capabilities: us - 0x8000e(GSM|ULAW|ALAW|H263), peer - audio=0x8(ALAW)/video=0x0(EMPTY), 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 192.168.0.206, port 5060 Transmitting: ACK sip:7001@192.168.0.206 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.200:5060;branch=z9hG4bK4799a5d2 From: "ATA Line 1" ;tag=as5539761f To: ;tag=464E90-1FE Contact: Call-ID: 3722563f6b547b605d2b9d6f5f1ad756@192.168.0.200 CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 192.168.0.206:5060 We're at 192.168.0.200 port 12878 Video is at 192.168.0.200 port 18752 Answering with capability 0x8(ALAW) Answering with non-codec capability 0x1(G723) Reliably Transmitting (no NAT): SIP/2.0 200 OK [Kasterisk*CLI> ia: SIP/2.0/UDP 192.168.0.201:5060 From: ;tag=545885975 To: ;tag=as4728682c Call-ID: 1683847546@192.168.0.201 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 218 o=root 25260 25261 IN IP4 192.168.0.200 s=session c=IN IP4 192.168.0.200 t=0 0 m=audio 12878 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - to 192.168.0.201:5060 [Kasterisk*CLI> Sip read: ACK sip:7001@192.168.0.200 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.201:5060 From: ;tag=545885975 To: ;tag=as4728682c Call-ID: 1683847546@192.168.0.201 CSeq: 1 ACK User-Agent: Cisco ATA 188 v3.1.1 atasip (040629A) Content-Length: 0 8 headers, 0 lines [Kasterisk*CLI> Sip read: BYE sip:2299@192.168.0.200:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.206:5060;branch=z9hG4bK1934 From: ;tag=464E90-1FE To: "ATA Line 1" ;tag=as5539761f Date: Fri, 01 Mar 2002 01:16:48 GMT Call-ID: 3722563f6b547b605d2b9d6f5f1ad756@192.168.0.200 User-Agent: Cisco-SIPGateway/IOS-12.x Max-Forwards: 70 Timestamp: 1014945412 CSeq: 101 BYE Reason: Q.850;cause=16 Content-Length: 0 12 headers, 0 lines Sending to 192.168.0.206 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.206:5060;branch=z9hG4bK1934 From: ;tag=464E90-1FE To: "ATA Line 1" ;tag=as5539761f Call-ID: 3722563f6b547b605d2b9d6f5f1ad756@192.168.0.200 CSeq: 101 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 192.168.0.206:5060 [Kasterisk*CLI> set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.0.201, port 5060 Reliably Transmitting: BYE sip:2299@192.168.0.201:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.200:5060;branch=z9hG4bK1445925b;rport From: ;tag=as4728682c To: ;tag=545885975 Contact: Call-ID: 1683847546@192.168.0.201 CSeq: 102 BYE User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 192.168.0.201:5060 [Kasterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.200:5060;branch=z9hG4bK1445925b;rport From: ;tag=as4728682c To: ;tag=545885975 Call-ID: 1683847546@192.168.0.201 CSeq: 102 BYE Server: Cisco ATA 188 v3.1.1 atasip (040629A) Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER Supported: replaces Content-Length: 0 10 headers, 0 lines Message is BYE Destroying call '3722563f6b547b605d2b9d6f5f1ad756@192.168.0.200' Destroying call '1683847546@192.168.0.201' [Kasterisk*CLI>