Audio is at 91.205.12.2 port 15856 Adding codec 0x8 (alaw) to SDP Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 83.169.182.1:5060: INVITE sip:0049865xxx7160@reg03.kabelphone.de;user=phone SIP/2.0 Via: SIP/2.0/UDP 91.205.12.2:5060;branch=z9hG4bK2519b2a9;rport Max-Forwards: 70 From: "0865xxxx3287" ;tag=as7631cfb8 To: Contact: Call-ID: 2bd5bcb87384e8492568951318eba732@reg03.kabelphone.de CSeq: 102 INVITE User-Agent: sip03.rsm-connect.net Date: Thu, 23 Apr 2009 19:02:44 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Type: application/sdp Content-Length: 283 v=0 o=root 568801415 568801415 IN IP4 91.205.12.2 s=Asterisk PBX 1.6.0.9 c=IN IP4 91.205.12.2 t=0 0 m=audio 15856 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv