Audio is at 19556
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x800000000000 (testlaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 81.31.45.51:5060:
INVITE sip:*41....@sip.odorik.cz SIP/2.0
Via: SIP/2.0/UDP 192.168.1.zzz:5060;branch=...;rport
Max-Forwards: 70
From: "dongle0" <sip:+420910xxxyyy@192.168.1.zzz>;tag=...
To: <sip:*41....@sip.odorik.cz>
Contact: <sip:+420910xxxyyy@192.168.1.zzz:5060>
Call-ID: 6ac3b1f94e...@192.168.1.zzz:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.13.1~dfsg-1
Date: Sat, 22 Dec 2012 09:31:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 291

v=0
o=root 1454605455 1454605455 IN IP4 192.168.1.zzz
s=Asterisk PBX 1.8.13.1~dfsg-1
c=IN IP4 192.168.1.zzz
t=0 0
m=audio 19556 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:81.31.45.51:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.zzz:5060;branch=z9hG...;rport=38349;received=moje-verejna-ip
From: "dongle0" <sip:+420910xxxyyy@192.168.1.zzz>;tag=as...
To: <sip:*41....@sip.odorik.cz>;tag=d9f55a....29ec
Call-ID: 6ac3b1...@192.168.1.zzz:5060
CSeq: 102 INVITE
Proxy-Authenticate: Digest realm="192.168.1.zzz", nonce="UNV/M1DV...=", qop="auth"
Server: Odorik UAS
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
set_destination: Parsing <sip:*41....@sip.odorik.cz> for address/port to send to
raspberrypi*CLI> 
==> /var/log/asterisk/messages <==
[Dec 22 10:31:52] NOTICE[2692] chan_sip.c: Failed to authenticate on INVITE to '"dongle0" <sip:+420910xxxyyy@192.168.1.zzz>;tag=...'

==> /var/log/asterisk/messages <==
[Dec 22 10:31:52] NOTICE[2692] chan_sip.c: Failed to authenticate on INVITE to '"dongle0" <sip:+420910xxxzzz@192.168.1.zzz>;tag=...'
set_destination: set destination to 81.31.45.51:5060
Transmitting (NAT) to 81.31.45.51:5060:
ACK sip:*41....@sip.odorik.cz SIP/2.0
Via: SIP/2.0/UDP 192.168.1.zzz:5060;branch=z9...;rport
Max-Forwards: 70
From: "dongle0" <sip:+420910xxxyyy@192.168.1.zzz>;tag=as...
To: <sip:*41....@sip.odorik.cz>;tag=d9f....29ec
Contact: <sip:+420910xxxyyy@192.168.1.zzz:5060>
Call-ID: ...6aa@192.168.1.zzz:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.13.1~dfsg-1
Content-Length: 0


---
[Dec 22 10:31:52] NOTICE[2692]: chan_sip.c:20309 handle_response_invite: Failed to authenticate on INVITE to '"dongle0" <sip:+420910xxxyyy@192.168.1.zzz>;tag=...'
Really destroying SIP dialog '6ac3...@192.168.1.zzz:5060' Method: INVITE