<--- SIP read from UDP:178.33.55.229:5060 --->
INVITE sip:6000@15.185.127.85 SIP/2.0
Via: SIP/2.0/UDP 178.33.55.229;rport;branch=z9hG4bK7S4D030c4Hm7F
Max-Forwards: 69
From: "Marcin Cieslak" <sip:saper@saper.info>;tag=DX83Z5HFB414m
To: <sip:6000@15.185.127.85>
Call-ID: 6e849723-91c4-1231-8cb2-3d6d4c7b9681
CSeq: 48909875 INVITE
Contact: <sip:mod_sofia@178.33.55.229:5060>
User-Agent: FreeSWITCH-mod_sofia/1.2.3
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
Supported: timer, precondition, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 279
X-FS-Support: update_display,send_info
Remote-Party-ID: "Marcin Cieslak" <sip:saper@saper.info>;party=calling;screen=yes;privacy=off

v=0
o=FreeSWITCH 489069639 489069640 IN IP4 178.33.55.229
s=FreeSWITCH
t=0 0
m=audio 29592 RTP/AVP 8 0 18 101
c=IN IP4 178.33.55.229
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (17 headers 12 lines) ---
Sending to 178.33.55.229:5060 (NAT)
Sending to 178.33.55.229:5060 (NAT)
Using INVITE request as basis request - 6e849723-91c4-1231-8cb2-3d6d4c7b9681
No matching peer for 'saper' from '178.33.55.229:5060'
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - (gsm|ulaw|alaw|g722), peer - audio=(ulaw|alaw|g729)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 178.33.55.229:29592
Looking for 6000 in public (domain 15.185.127.85)
list_route: hop: <sip:mod_sofia@178.33.55.229:5060>

<--- Transmitting (NAT) to 178.33.55.229:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 178.33.55.229;branch=z9hG4bK7S4D030c4Hm7F;received=178.33.55.229;rport=5060
From: "Marcin Cieslak" <sip:saper@saper.info>;tag=DX83Z5HFB414m
To: <sip:6000@15.185.127.85>
Call-ID: 6e849723-91c4-1231-8cb2-3d6d4c7b9681
CSeq: 48909875 INVITE
Server: Asterisk PBX 11.5.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:6000@10.5.203.70:5060>
Content-Length: 0


<------------>
Audio is at 14034
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to 178.33.55.229:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 178.33.55.229;branch=z9hG4bK7S4D030c4Hm7F;received=178.33.55.229;rport=5060
From: "Marcin Cieslak" <sip:saper@saper.info>;tag=DX83Z5HFB414m
To: <sip:6000@15.185.127.85>;tag=as2c86ef02
Call-ID: 6e849723-91c4-1231-8cb2-3d6d4c7b9681
CSeq: 48909875 INVITE
Server: Asterisk PBX 11.5.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:6000@10.5.203.70:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 257

v=0
o=root 1048923561 1048923561 IN IP4 10.5.203.70
s=Asterisk PBX 11.5.1
c=IN IP4 10.5.203.70
t=0 0
m=audio 14034 RTP/AVP 0 8 101
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

<------------>
Retransmitting #1 (NAT) to 178.33.55.229:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 178.33.55.229;branch=z9hG4bK7S4D030c4Hm7F;received=178.33.55.229;rport=5060
From: "Marcin Cieslak" <sip:saper@saper.info>;tag=DX83Z5HFB414m
To: <sip:6000@15.185.127.85>;tag=as2c86ef02
Call-ID: 6e849723-91c4-1231-8cb2-3d6d4c7b9681
CSeq: 48909875 INVITE
Server: Asterisk PBX 11.5.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:6000@10.5.203.70:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 257

v=0
o=root 1048923561 1048923561 IN IP4 10.5.203.70
s=Asterisk PBX 11.5.1
c=IN IP4 10.5.203.70
t=0 0
m=audio 14034 RTP/AVP 0 8 101
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

---