phone2*CLI> We're at 10.15.1.2 port 19196 Video is at 10.15.1.2 port 10528 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x80000 (h263) to SDP Adding non-codec 0x1 (telephone-event) to SDP 13 headers, 14 lines Reliably Transmitting (NAT) to 10.15.4.254:5060: INVITE sip:1125@10.15.4.254:5060 SIP/2.0 Via: SIP/2.0/UDP 10.15.1.2:5060;branch=z9hG4bK2f51d1c6;rport From: "4951293436" ;tag=as09cc3131 To: Contact: Call-ID: 3fd0e0e973fae7b1001acca6356235d6@masterhost.ru CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 06 Jul 2006 06:51:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 305 v=0 o=root 9411 9411 IN IP4 10.15.1.2 s=session c=IN IP4 10.15.1.2 t=0 0 m=audio 19196 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - m=video 10528 RTP/AVP 34 a=rtpmap:34 H263/90000 --- Destroying call '3fd0e0e973fae7b1001acca6356235d6@masterhost.ru' phone2*CLI> <-- SIP read from 10.15.4.254:5060: SIP/2.0 100 Trying Call-ID: 3fd0e0e973fae7b1001acca6356235d6@masterhost.ru CSeq: 102 INVITE From: "4951293436" ;tag=as09cc3131 To: ;tag=59093062c15e0b4 Via: SIP/2.0/UDP 10.15.1.2:5060;branch=z9hG4bK2f51d1c6;rport Content-Length: 0 User-Agent: Avaya SIP R2.2 Endpoint Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 --- (8 headers 0 lines)--- phone2*CLI> Destroying call '3fd0e0e973fae7b1001acca6356235d6@masterhost.ru' phone2*CLI> <-- SIP read from 10.15.4.254:5060: SIP/2.0 180 Ringing Call-ID: 3fd0e0e973fae7b1001acca6356235d6@masterhost.ru CSeq: 102 INVITE From: "4951293436" ;tag=as09cc3131 To: ;tag=59093062c15e0b4 Via: SIP/2.0/UDP 10.15.1.2:5060;branch=z9hG4bK2f51d1c6;rport Content-Length: 0 Contact: User-Agent: Avaya SIP R2.2 Endpoint Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 --- (9 headers 0 lines)--- Destroying call '3fd0e0e973fae7b1001acca6356235d6@masterhost.ru'