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" <sip:0865xxxx3287@reg03.kabelphone.de>;tag=as7631cfb8
To: <sip:0049865xxx7160@reg03.kabelphone.de;user=phone>
Contact: <sip:0865xxxx3287@91.205.12.2>
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