--- (16 headers 20 lines) ---
Sending to 193.93.205.34 : 5060 (no NAT)
Using INVITE request as basis request - 5e13c2484c2e3d7316388ea01bbba7ca@213.199.197.30
Found peer 'easycall3'
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 0
Found RTP audio format 3
Found RTP audio format 97
Found RTP audio format 101
Peer audio RTP is at port 193.93.205.34:59250
Found description format PCMA for ID 8
Found description format G729 for ID 18
Found description format PCMU for ID 0
Found description format GSM for ID 3
Found description format iLBC for ID 97
Found description format telephone-event for ID 101
Capabilities: us - 0x4 (ulaw), peer - audio=0x50e (gsm|ulaw|alaw|g729|ilbc)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 193.93.205.34:59250
Looking for 6000 in stations-mind (domain 192.168.127.253)
<--- Reliably Transmitting (no NAT) to 193.93.205.34:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 193.93.205.34;branch=z9hG4bK7525.247a2212.0;received=193.93.205.34
Via: SIP/2.0/UDP 213.199.197.30:5060;branch=z9hG4bK6221348b;rport=5060
From: "0226303272" <sip:0226303272@213.199.197.30>;tag=as1ff40c90
To: <sip:990193@sip.easycall.pl>;tag=as424e57e8
Call-ID: 5e13c2484c2e3d7316388ea01bbba7ca@213.199.197.30
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '5e13c2484c2e3d7316388ea01bbba7ca@213.199.197.30' in 32000 ms (Method: INVITE)
<--- SIP read from 193.93.205.34:5060 --->
ACK sip:6000@192.168.127.253:5060 SIP/2.0
Via: SIP/2.0/UDP 193.93.205.34;branch=z9hG4bK7525.247a2212.0
From: "0226303272" <sip:0226303272@213.199.197.30>;tag=as1ff40c90
Call-ID: 5e13c2484c2e3d7316388ea01bbba7ca@213.199.197.30
To: <sip:990193@sip.easycall.pl>;tag=as424e57e8
CSeq: 102 ACK
Content-Length: 0