== Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Found Asterisk CVS-HEAD-12/28/04-18:37:46, Copyright (C) 1999-2004 Digium. Written by Mark Spencer ========================================================================= Connected to Asterisk CVS-HEAD-12/28/04-18:37:46 currently running on astlap (pid = 22128) astlap*CLI> Usage: set verbose Sets level of verbose messages to be displayed. 0 means no messages should be displayed. Equivalent to -v[v[v...]] on startup Usage: set debug Sets level of core debug messages to be displayed. 0 means no messages should be displayed. Equivalent to -d[d[d...]] on startup. astlap*CLI> sip debug astlap*CLI> SIP Debugging Enabled astlap*CLI> Sip read: 0 headers, 0 lines astlap*CLI> Sip read: INVITE sip:256222@67.165.241.16 SIP/2.0 Max-Forwards: 10 Record-Route: Via: SIP/2.0/UDP 69.90.155.70;branch=z9hG4bKc896.36478dd3.0 Via: SIP/2.0/UDP 66.54.140.46:5060;branch=z9hG4bK4877ad14 From: "9704020652" ;tag=as2b4e1dd4 To: Contact: Call-ID: 510f73096a4d50bf50f3fdb478bc32f3@66.54.140.46 CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Wed, 29 Dec 2004 03:11:28 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 263 v=0 o=root 31243 31243 IN IP4 66.54.140.46 s=session c=IN IP4 66.54.140.46 t=0 0 m=audio 14372 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 a=silenceSupp:off - - - - 15 headers, 12 lines Using latest request as basis request Sending to 69.90.155.70 : 5060 (non-NAT) Found RTP audio format 3 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 66.54.140.46:14372 Found description format GSM Found description format PCMU Found description format PCMA Found description format telephone-event Capabilities: us - 0x116 (gsm|ulaw|g726|g729), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing), combined - 0x6 (gsm|ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) astlap*CLI> Found no matching peer or user for '69.90.155.70:5060' Looking for 256222 in from-direct-ip Reliably Transmitting (no NAT): SIP/2.0 404 Not Found Via: SIP/2.0/UDP 69.90.155.70;branch=z9hG4bKc896.36478dd3.0 Via: SIP/2.0/UDP 66.54.140.46:5060;branch=z9hG4bK4877ad14 From: "9704020652" ;tag=as2b4e1dd4 To: ;tag=as52b885ae Call-ID: 510f73096a4d50bf50f3fdb478bc32f3@66.54.140.46 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 69.90.155.70:5060 astlap*CLI> Sip read: ACK sip:256222@67.165.241.16 SIP/2.0 Via: SIP/2.0/UDP 69.90.155.70;branch=z9hG4bKc896.36478dd3.0 From: "9704020652" ;tag=as2b4e1dd4 Call-ID: 510f73096a4d50bf50f3fdb478bc32f3@66.54.140.46 To: ;tag=as52b885ae CSeq: 102 ACK User-Agent: Sip EXpress router(0.8.14 (i386/linux)) Content-Length: 0 8 headers, 0 lines Destroying call '510f73096a4d50bf50f3fdb478bc32f3@66.54.140.46' astlap*CLI> sip no debugshow pe peer peers astlap*CLI> sip show peer fwd astlap*CLI> * Name : fwd Secret : MD5Secret : Context : from-sip Language : en FromUser : 256222 FromDomain : fwd.pulver.com Callgroup : (0) Pickupgroup : (0) Mailbox : LastMsgsSent : -1 Dynamic : No Callerid : "" <> Expire : -1 Expiry : 900 Insecure : Very Nat : No ACL : No CanReinvite : Yes PromiscRedir : No User=Phone : No DTMFmode : rfc2833 LastMsg : 0 ToHost : fwd.pulver.com Addr->IP : 69.90.155.70 Port 5060 Defaddr->IP : 0.0.0.0 Port 0 Username : 256222 Codecs : 0x4 (ulaw) Codec Order : (ulaw) Status : UNKNOWN Useragent : Full Contact :