pbx*CLI> pbx*CLI> sip show peers Name/username Host Dyn Nat ACL Port Status sip_proxy-out/gustavofeli 194.221.62.207 5060 OK (52 ms) raquel (Unspecified) D N 0 UNKNOWN gs/gs 212.18.170.170 D 5060 OK (37 ms) ekiga/ekiga 212.18.170.52 D 5062 OK (31 ms) 1234/1234 (Unspecified) D 0 UNKNOWN 5 sip peers [3 online , 2 offline] pbx*CLI> pbx*CLI> pbx*CLI> pbx*CLI> pbx*CLI> pbx*CLI> pbx*CLI> pbx*CLI> pbx*CLI> pbx*CLI> pbx*CLI> sip debug SIP Debugging enabled pbx*CLI> sip show peers Name/username Host Dyn Nat ACL Port Status sip_proxy-out/gustavofeli 194.221.62.207 5060 OK (55 ms) raquel (Unspecified) D N 0 UNKNOWN gs/gs 212.18.170.170 D 5060 OK (33 ms) ekiga/ekiga 212.18.170.52 D 5062 OK (30 ms) 1234/1234 (Unspecified) D 0 UNKNOWN 5 sip peers [3 online , 2 offline] pbx*CLI> <-- SIP read from 212.18.170.52:5078: INVITE sip:gs@pbx.dei.uc.pt SIP/2.0 Route: Date: Mon, 06 Nov 2006 16:35:47 GMT CSeq: 1 INVITE Via: SIP/2.0/UDP 212.18.170.52:5078;branch=z9hG4bK44057986-226c-db11-9a69-000c76f92fd8;rport User-Agent: Ekiga/2.0.2 From: "Gustavo Felisberto" ;tag=b4277886-226c-db11-9a69-000c76f92fd8 Call-ID: 4e247886-226c-db11-9a69-000c76f92fd8@sam To: Contact: Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,NOTIFY,REFER,MESSAGE Content-Type: application/sdp Content-Length: 451 Max-Forwards: 70 v=0 o=- 1162830947 1162830947 IN IP4 212.18.170.52 s=Opal SIP Session c=IN IP4 212.18.170.52 t=0 0 m=audio 5004 RTP/AVP 101 114 115 3 107 110 0 8 106 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtpmap:114 SPEEX/16000 a=rtpmap:115 iLBC/8000 a=rtpmap:3 GSM/8000 a=rtpmap:107 MS-GSM/8000 a=rtpmap:110 SPEEX/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 G726-16k/8000 m=video 5006 RTP/AVP 31 a=rtpmap:31 H261/90000 --- (14 headers 18 lines) --- Using INVITE request as basis request - 4e247886-226c-db11-9a69-000c76f92fd8@sam Sending to 212.18.170.52 : 5078 (NAT) Found user 'ekiga' Found RTP audio format 101 Found RTP audio format 114 Found RTP audio format 115 Found RTP audio format 3 Found RTP audio format 107 Found RTP audio format 110 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 106 Found RTP video format 31 Peer audio RTP is at port 212.18.170.52:5004 Peer video RTP is at port 212.18.170.52:5006 Found description format telephone-event Found description format SPEEX Found description format iLBC Found description format GSM Found description format MS-GSM Found description format SPEEX Found description format PCMU Found description format PCMA Found description format G726-16k Found description format H261 Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x60e (gsm|ulaw|alaw|speex|ilbc)/video=0x40000 (h261), combined - 0xe (gsm|ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for gs in default (domain pbx.dei.uc.pt) Reliably Transmitting (no NAT) to 212.18.170.52:5078: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 212.18.170.52:5078;branch=z9hG4bK44057986-226c-db11-9a69-000c76f92fd8;rport;received=212.18.170.52 From: "Gustavo Felisberto" ;tag=b4277886-226c-db11-9a69-000c76f92fd8 To: ;tag=as336dd5e9 Call-ID: 4e247886-226c-db11-9a69-000c76f92fd8@sam CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- pbx*CLI> <-- SIP read from 212.18.170.52:5078: ACK sip:gs@pbx.dei.uc.pt SIP/2.0 Route: CSeq: 1 ACK Via: SIP/2.0/UDP 212.18.170.52:5078;branch=z9hG4bK44057986-226c-db11-9a69-000c76f92fd8;rport From: "Gustavo Felisberto" ;tag=b4277886-226c-db11-9a69-000c76f92fd8 Call-ID: 4e247886-226c-db11-9a69-000c76f92fd8@sam To: ;tag=as336dd5e9 Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,NOTIFY,REFER,MESSAGE Content-Length: 0 Max-Forwards: 70 --- (10 headers 0 lines) --- Destroying call '4e247886-226c-db11-9a69-000c76f92fd8@sam' pbx*CLI> pbx ~ #