SIP Phone> show regi LINE REGISTRATION TABLE Proxy Registration: ENABLED, state: REGISTERED line APR state timer expires proxy:port ---- --- ------------- ---------- ---------- ---------------------------- 1 111 REGISTERED 3595 2918 192.168.100.1:5060 2 ... NONE 0 0 undefined:0 3 ... NONE 0 0 undefined:0 4 ... NONE 0 0 undefined:0 5 ... NONE 0 0 undefined:0 6 ... NONE 0 0 undefined:0 1-BU .1x IDLE 0 0 undefined:0 Note: APR is Authenticated, Provisioned, Registered

SIP Phone> register 1 Incomplete command reg [option] [line] options = 0: unregister 1: register line = 1 through 6 = backup (line 1 to backup proxy) SIP Phone> register 1 1 Registering line 1 SIP Phone> exit

SIP Phone> test ? Test Command Definitions ------------------------ onhook , hu - Handset Onhook offhook , hd - Handset Offhook key , ky - Simulate Keystrokes open , op - Open the Test Session close , cl - Close the Test Session show , sh - Show Call Feedback hide , hi - Hide Call Feedback SIP Phone> test op TEST: Opening Session SIP Phone> test key Test Key Names -------------- 0-9 # * line1 line2 navup navdn volup voldn soft1 soft2 soft3 soft4 serv info dir msgs set headset spkr mute SIP Phone> test key set SIP Phone> test key 6 SIP Phone> test key 5 SIP Phone> test key soft1 SIP Phone> test key soft2 SIP Phone> test cl

SIP Phone> debug ? console-stall malloc-table sk-platform flash http timestamp timers dsp dsp-keepalive vcm dtmf cpr-info task-socket xml-events xml-deck xml-vars xml-post lsm fsm auth fim gsm cc cc-msg sip-task sip-state sip-messages sip-reg-state sip-trx cdp dns sntp sntp-packet config debugs: NONE

SIP Phone> debug sip-messages Enabling bug logging on this terminal - use 'tty mon 0' to disable debugs: timestamp sip-messages SIP Phone> [13:02:28:68479615] SIPProcessUDPMessage: recv UDP message from <192.168.100.1>:<50195>, length=<564>, message= [13:02:28:68479615] OPTIONS sip:1006@192.168.100.1:5061;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.100.1:5060;branch=z9hG4bK62bb97a3;rport Max-Forwards: 70 From: "asterisk" ;tag=as3db05137 To: Contact: Call-ID: 539f88c326a934223f3b2fd058d3d047@192.168.100.1 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.18 Date: Sat, 25 Jun 2011 12:48:13 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 [13:02:28:68479621] sippmh_parse_supported_require: Invalid tag in Require/Supported timer [13:02:28:68479625] sipRelDevCoupledMessageStore: Storing for reTx (cseq=102, method=OPTIONS, to_tag=<001646d29adc2a3159982181-2319c0e4>) [13:02:28:68479627] sipTransportSendMessage: Opened a one-time UDP send channel to server <192.168.100.1>:<5060>, handle = 14 local port= 5061 [13:02:28:68479628] sipTransportSendMessage:Sent SIP message to <192.168.100.1>:<5060>, handle=<14>, length=<912>, message= [13:02:28:68479628] SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.100.1:5060;branch=z9hG4bK62bb97a3;rport From: "asterisk" ;tag=as3db05137 To: ;tag=001646d29adc2a3159982181-2319c0e4 Call-ID: 539f88c326a934223f3b2fd058d3d047@192.168.100.1 CSeq: 102 OPTIONS Server: Cisco-CP7940G/8.0 Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE Accept: application/sdp,multipart/mixed,multipart/alternative Accept-Encoding: identity Accept-Language: en Supported: replaces,join,norefersub Content-Length: 261 Content-Type: application/sdp Content-Disposition: session;handling=optional v=0 o=Cisco-SIPUA 6506 0 IN IP4 192.168.100.1 s=SIP Call t=0 0 m=audio 0 RTP/AVP 0 8 13 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:13 CN/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 [13:02:28:68479630] sipTransportSendMessage: Closed a one-time UDP send channel handle = 14