[root@pbx asterisk]# asterisk -vvvvvvvvvvvvvvvvvvvvddddddddddddddddddrgc Parsing /etc/asterisk/asterisk.conf Asterisk 1.2.10, Copyright (C) 1999 - 2006 Digium, Inc. and others. Created by Mark Spencer Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'show license' for details. ========================================================================= Connected to Asterisk 1.2.10 currently running on pbx (pid = 26872) Verbosity was 3 and is now 20 Core debug was 0 and is now 18 Asterisk Ready. -- Remote UNIX connection pbx*CLI> pbx*CLI> sip debug peer 207.218.174.244 SIP Debugging Enabled for IP: 207.218.174.244:5060 pbx*CLI> <-- SIP read from 207.218.174.244:5060: INVITE sip:17135745600;npdi=yes;rn=17135745600@XXX.XXX.XXX.XXX:5060 SIP/2.0 Via: SIP/2.0/UDP 207.218.174.244:5060;branch=z9hG4bK00252m100gm1skc70641.1 From: ;tag=SDs46a201-gK0d5ba019 To: Call-ID: SDs46a201-b433483c00496a6bbd048565fcfd1e8a-v3000i1 CSeq: 31123 INVITE Max-Forwards: 69 Allow: OPTIONS,INVITE,ACK,CANCEL,BYE,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE Accept: multipart/mixed, application/sdp, application/isup, application/dtmf, application/dtmf-relay Contact: Content-Length: 186 Content-Disposition: session; handling=required Content-Type: application/sdp P-Asserted-Identity: v=0 o=Sonus_UAC 22906 26156 IN IP4 207.218.174.244 s=SIP Media Capabilities c=IN IP4 207.218.174.244 t=0 0 m=audio 53174 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=sendrecv a=maxptime:20 --- (14 headers 9 lines)--- Using INVITE request as basis request - SDs46a201-b433483c00496a6bbd048565fcfd1e8a-v3000i1 Sending to 207.218.174.244 : 5060 (NAT) Found peer '207.218.174.244' Found RTP audio format 0 Peer audio RTP is at port 207.218.174.244:53174 Found description format PCMU Capabilities: us - 0x51f (g723|gsm|ulaw|alaw|g726|g729|ilbc), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Looking for s in global (domain 17135745600) Reliably Transmitting (NAT) to 207.218.174.244:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 207.218.174.244:5060;branch=z9hG4bK00252m100gm1skc70641.1;received=207.218.174.244 From: ;tag=SDs46a201-gK0d5ba019 To: ;tag=as785b49c2 Call-ID: SDs46a201-b433483c00496a6bbd048565fcfd1e8a-v3000i1 CSeq: 31123 INVITE User-Agent: pbx Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- pbx*CLI> <-- SIP read from 207.218.174.244:5060: ACK sip:17135745600;npdi=yes;rn=17135745600@XXX.XXX.XXX.XXX:5060 SIP/2.0 Via: SIP/2.0/UDP 207.218.174.244:5060;branch=z9hG4bK00252m100gm1skc70641.1 CSeq: 31123 ACK From: ;tag=SDs46a201-gK0d5ba019 To: ;tag=as785b49c2 Call-ID: SDs46a201-b433483c00496a6bbd048565fcfd1e8a-v3000i1 Max-Forwards: 69 Content-Length: 0 --- (8 headers 0 lines)--- Destroying call 'SDs46a201-b433483c00496a6bbd048565fcfd1e8a-v3000i1' pbx*CLI> exit Executing last minute cleanups Asterisk cleanly ending (0). [root@pbx asterisk]#