INVITE sip:17183301234@203.160.242.66 SIP/2.0 Via: SIP/2.0/UDP 202.65.218.19:5060;branch=z9hG4bK02ba8818;rport From: "0136332233" ;tag=as63c981b3 To: Contact: Call-ID: 5713741a4f11288b5f8360d80d951e98@202.65.218.19 CSeq: 102 INVITE User-Agent: ASTERISK Max-Forwards: 70 Remote-Party-ID: "0136332233" ;privacy=off;screen=no Date: Tue, 21 Nov 2006 17:47:45 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 263 v=0 o=root 26078 26078 IN IP4 202.65.218.19 s=session c=IN IP4 202.65.218.19 t=0 0 m=audio 22800 RTP/AVP 4 0 8 101 a=rtpmap:4 G723/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Nov 22 01:47:45] -- Called N-IDD/17183301234 [Nov 22 01:47:45] <--- SIP read from 203.160.242.66:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 202.65.218.19:5060;branch=z9hG4bK02ba8818;rport From: "0136332233" ;tag=as63c981b3 To: ;tag=A77CACDC-AD7 Date: Tue, 21 Nov 2006 17:47:46 GMT Call-ID: 5713741a4f11288b5f8360d80d951e98@202.65.218.19 Server: Cisco-SIPGateway/IOS-12.x CSeq: 102 INVITE Allow-Events: telephone-event Content-Length: 0 <-------------> [Nov 22 01:47:45] --- (10 headers 0 lines) --- [Nov 22 01:47:45] <--- SIP read from 203.160.242.66:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 202.65.218.19:5060;branch=z9hG4bK02ba8818;rport From: "0136332233" ;tag=as63c981b3 To: ;tag=A77CACDC-AD7 Date: Tue, 21 Nov 2006 17:47:46 GMT Call-ID: 5713741a4f11288b5f8360d80d951e98@202.65.218.19 Server: Cisco-SIPGateway/IOS-12.x CSeq: 102 INVITE Allow-Events: telephone-event Contact: Content-Disposition: session;handling=required Content-Type: application/sdp Content-Length: 261 v=0 o=CiscoSystemsSIP-GW-UserAgent 3235 5767 IN IP4 203.160.242.66 s=SIP Call c=IN IP4 203.160.242.66 t=0 0 m=audio 17002 RTP/AVP 4 101 c=IN IP4 203.160.242.66 a=rtpmap:4 G723/8000 a=fmtp:4 annexa=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 <-------------> [Nov 22 01:47:45] --- (13 headers 11 lines) --- [Nov 22 01:47:45] Found RTP audio format 4 [Nov 22 01:47:45] Found RTP audio format 101 [Nov 22 01:47:45] Peer audio RTP is at port 203.160.242.66:17002 [Nov 22 01:47:45] Found description format G723 for ID 4 [Nov 22 01:47:45] Found description format telephone-event for ID 101 [Nov 22 01:47:45] Capabilities: us - 0xd (g723|ulaw|alaw), peer - audio=0x1 (g723)/video=0x0 (nothing), combined - 0x1 (g723) [Nov 22 01:47:45] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Nov 22 01:47:45] Peer audio RTP is at port 203.160.242.66:17002 [Nov 22 01:47:45] -- SIP/N-IDD-006ca240 is making progress passing it to SIP/607001100058-006c3d70 [Nov 22 01:47:49] <--- SIP read from 203.160.242.66:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 202.65.218.19:5060;branch=z9hG4bK02ba8818;rport From: "0136332233" ;tag=as63c981b3 To: ;tag=A77CACDC-AD7 Date: Tue, 21 Nov 2006 17:47:46 GMT Call-ID: 5713741a4f11288b5f8360d80d951e98@202.65.218.19 Server: Cisco-SIPGateway/IOS-12.x CSeq: 102 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO Allow-Events: telephone-event Contact: Content-Type: application/sdp Content-Length: 261 v=0 o=CiscoSystemsSIP-GW-UserAgent 3235 5767 IN IP4 203.160.242.66 s=SIP Call c=IN IP4 203.160.242.66 t=0 0 m=audio 17002 RTP/AVP 4 101 c=IN IP4 203.160.242.66 a=rtpmap:4 G723/8000 a=fmtp:4 annexa=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 <-------------> [Nov 22 01:47:49] --- (13 headers 11 lines) --- [Nov 22 01:47:49] Found RTP audio format 4 [Nov 22 01:47:49] Found RTP audio format 101 [Nov 22 01:47:49] Peer audio RTP is at port 203.160.242.66:17002 [Nov 22 01:47:49] Found description format G723 for ID 4 [Nov 22 01:47:49] Found description format telephone-event for ID 101 [Nov 22 01:47:49] Capabilities: us - 0xd (g723|ulaw|alaw), peer - audio=0x1 (g723)/video=0x0 (nothing), combined - 0x1 (g723) [Nov 22 01:47:49] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Nov 22 01:47:49] Peer audio RTP is at port 203.160.242.66:17002 [Nov 22 01:47:49] list_route: hop: [Nov 22 01:47:49] set_destination: Parsing for address/port to send to [Nov 22 01:47:49] set_destination: set destination to 203.160.242.66, port 5060 [Nov 22 01:47:49] Transmitting (no NAT) to 203.160.242.66:5060: ACK sip:17183301234@203.160.242.66:5060 SIP/2.0 Via: SIP/2.0/UDP 202.65.218.19:5060;branch=z9hG4bK1d88f16b;rport From: "0136332233" ;tag=as63c981b3 To: ;tag=A77CACDC-AD7 Contact: Call-ID: 5713741a4f11288b5f8360d80d951e98@202.65.218.19 CSeq: 102 ACK User-Agent: ASTERISK Max-Forwards: 70 Remote-Party-ID: "0136332233" ;privacy=off;screen=no Content-Length: 0 --- [Nov 22 01:47:49] -- SIP/N-IDD-006ca240 answered SIP/607001100058-006c3d70 [Nov 22 01:47:49] -- Packet2Packet bridging SIP/607001100058-006c3d70 and SIP/N-IDD-006ca240 [Nov 22 01:47:52] Scheduling destruction of SIP dialog '5713741a4f11288b5f8360d80d951e98@202.65.218.19' in 32000 ms (Method: INVITE) [Nov 22 01:47:52] set_destination: Parsing for address/port to send to [Nov 22 01:47:52] set_destination: set destination to 203.160.242.66, port 5060 [Nov 22 01:47:52] Reliably Transmitting (no NAT) to 203.160.242.66:5060: BYE sip:17183301234@203.160.242.66:5060 SIP/2.0 Via: SIP/2.0/UDP 202.65.218.19:5060;branch=z9hG4bK55c6585b;rport From: "0136332233" ;tag=as63c981b3 To: ;tag=A77CACDC-AD7 Call-ID: 5713741a4f11288b5f8360d80d951e98@202.65.218.19 CSeq: 103 BYE User-Agent: ASTERISK Max-Forwards: 70 Remote-Party-ID: "0136332233" ;privacy=off;screen=no Content-Length: 0 <--- SIP read from 203.160.242.66:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 202.65.218.19:5060;branch=z9hG4bK55c6585b;rport From: "0136332233" ;tag=as63c981b3 To: ;tag=A77CACDC-AD7 Date: Tue, 21 Nov 2006 17:47:53 GMT Call-ID: 5713741a4f11288b5f8360d80d951e98@202.65.218.19 Server: Cisco-SIPGateway/IOS-12.x Content-Length: 0 CSeq: 103 BYE