 give you an idea of how verbose SIP can be, here's the output from me dialing a phone number from my softphone, then hanging up before there's an answer: 570 Chapter 30: VolP *Feb 3 01:17:07.740: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Sent: INVITE sip:9735554471@sbc.sip-provider.com:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.3:5060;branch=z9hG4bKB176: From: <sip:9735551212@sbc.sip-provider.com>;tag=383B4-0 To: <sip:9735554471@sbc.sip-provider.com> Date: Thu, 03 Feb 2011 01:17:07 GMT Call-ID: 26F3EF99-2E6A11E0-800D80A2-733EA81A@192.168.0.3 Supported: 100rel,timer,resource-priority,replaces,sdp-anat Min-SE: 1800 Cisco-Guid: 0563478376-0778703328-2148040866-193348610 User-Agent: Cisco-SIPGateway/I0S-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER CSeq: 102 INVITE Max-Forwards: 70 Timestamp: 1296695827 Contact: <sip:9735551212@192.168.0.3:5060> Expires: 180 Allow-Events: telephone-event Proxy-Authorization: Digest username="9735551212",realm="voip.sip-provider.com' uri="sip:9735554471@sbc.sip-provider.com:5060", response="48fab7d8750961aa91e43f11879ddd4f",nonce="9967d6506474325b254afb6d6bc1baf4", algorithm=md5 Content-Type: application/sdp Content-Disposition: session;handling=required Content-Length: 264 o=CiscoSystemsSIP-GW-UserAgent 8383 1800 IN IP4 192.168.0.3 s=SIP Call c=IN IP4 192.168.0.3 m=audio 18674 RTP/AVP 0 101 c=IN IP4 192.168.0.3 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=direction:active *Feb 3 01:17:07.844: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Received: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.3:5060;branch=z9hG4bKB1761 From: sip:9735551212@sbc.sip-provider.com>;tag=383B4-0 To: ip:9735554471@sbc.sip-provider.com> Call-ID: 26F3EF99-2E6A11E0-800D80A2-733EA81A@192.168.0.3 CSeq: 102 INVITE Timestamp: 1296695827 *Feb 3 01:17:11.004: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Sent: CANCEL sip:9735551212@sbc.sip-provider.com:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.3:5060;branch=z9hG4bKB1761 Troublesho