<-- SIP read from 212.227.15.197:5060:
INVITE sip:[email protected] SIP/2.0
Record-Route: <sip:212.227.15.198;ftag=0F4565BA3C56D591;lr=on>
Via: SIP/2.0/UDP 212.227.15.197;branch=z9hG4bKa3573255e7a789d5475a5098990486e1
Via: SIP/2.0/UDP 212.227.15.198;branch=z9hG4bK1724.eb328413.0
Via: SIP/2.0/UDP 212.227.15.197;branch=z9hG4bKc50902f204c4256bfeba2b8efdc21a41
Via: SIP/2.0/UDP 84.169.163.125:5060;branch=z9hG4bKCAEDEDB89C5F3EB2
From: <sip:[email protected]>;tag=0F4565BA3C56D591
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 653 INVITE
Contact: <sip:[email protected];uniq=F2B6A07EE2E427FDA1C22F6C6D5BB>
Max-Forwards: 14
Expires: 120
User-Agent: AVM FRITZ!Box Fon WLAN 7050 (UI) 14.04.15 (Jul 12 2006)
Supported: 100rel,replaces,timer
Allow-Events: telephone-event,refer
Allow: INVITE,ACK,OPTIONS,CANCEL,BYE,UPDATE,PRACK,INFO,SUBSCRIBE,NOTIFY,REFER,MESSAGE
Content-Type: application/sdp
Accept: application/sdp, multipart/mixed
Accept-Encoding: identity
Content-Length: 379
v=0
o=user 5476713 5476713 IN IP4 84.169.163.125
s=call
c=IN IP4 84.169.163.125
t=1159791143 1159794743
m=audio 7078 RTP/AVP 8 0 2 102 100 99 97 101
a=sendrecv
a=rtpmap:2 G726-32/8000
a=rtpmap:102 G726-32/8000
a=rtpmap:100 G726-40/8000
a=rtpmap:99 G726-24/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
a=rtcp:7079
--- (21 headers 16 lines)---
Using INVITE request as basis request - [email protected]
Sending to 212.227.15.197 : 5060 (non-NAT)
Found peer '496144xxxxxx'
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 2
Found RTP audio format 102
Found RTP audio format 100
Found RTP audio format 99
Found RTP audio format 97
Found RTP audio format 101
Peer audio RTP is at port 84.169.163.125:7078
Found description format G726-32
Found description format G726-32
Found description format G726-40
Found description format G726-24
Found description format iLBC
Found description format telephone-event
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x41c (ulaw|alaw|g726|ilbc)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for 496144xxxxxx in default (domain 12.12.123.12)
Reliably Transmitting (no NAT) to 212.227.15.197:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 212.227.15.197;branch=z9hG4bKa3573255e7a789d5475a5098990486e1;received=212.227.15.197
Via: SIP/2.0/UDP 212.227.15.198;branch=z9hG4bK1724.eb328413.0
Via: SIP/2.0/UDP 212.227.15.197;branch=z9hG4bKc50902f204c4256bfeba2b8efdc21a41
Via: SIP/2.0/UDP 84.169.163.125:5060;branch=z9hG4bKCAEDEDB89C5F3EB2
From: <sip:[email protected]>;tag=0F4565BA3C56D591
To: <sip:[email protected]>;tag=as7bb47930
Call-ID: [email protected]
CSeq: 653 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:[email protected]>
Content-Length: 0
---
srv02*CLI>
<-- SIP read from 212.227.15.197:5060:
ACK sip:[email protected] SIP/2.0
Max-Forwards: 10
Via: SIP/2.0/UDP 212.227.15.197;branch=z9hG4bKa3573255e7a789d5475a5098990486e1
Via: SIP/2.0/UDP 212.227.15.198;branch=z9hG4bK1724.eb328413.0
From: <sip:[email protected]>;tag=0F4565BA3C56D591
Call-ID: [email protected]
To: <sip:[email protected]>;tag=as7bb47930
CSeq: 653 ACK
User-Agent: UI OpenSer
Content-Length: 0
--- (10 headers 0 lines)---
Destroying call '[email protected]'