<--- SIP read from UDP:192.168.178.1:5060 --->
INVITE sip:666@192.168.178.21 SIP/2.0
Via: SIP/2.0/UDP 192.168.178.1:5060;rport;branch=z9hG4bK8483D720AD7EC31A
Route: <sip:192.168.178.21;lr>
From: <sip:test@192.168.178.21>;tag=810E009F9191EF51
To: <sip:666@192.168.178.21>
Call-ID: 049B62F3698D53D8@192.168.178.1
CSeq: 27 INVITE
Contact: <sip:test@192.168.178.1;uniq=80DE750FD696A6F05E4C556872D86>
Max-Forwards: 70
Expires: 120
User-Agent: AVM FRITZ!Box Fon WLAN 7360 111.05.24 (Jul 6 2012)
Supported: 100rel,replaces
Allow-Events: telephone-event,refer
Allow: INVITE,ACK,OPTIONS,CANCEL,BYE,UPDATE,PRACK,INFO,SUBSCRIBE,NOTIFY,REFER,MESSAGE,PUBLISH
Content-Type: application/sdp
Accept: application/sdp, multipart/mixed
Accept-Encoding: identity
Content-Length: 395

v=0
o=user 2529382 2529382 IN IP4 192.168.178.1
s=call
c=IN IP4 192.168.178.1
t=0 0
m=audio 7078 RTP/AVP 8 0 2 102 100 99 97 18 101
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=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtcp:7079
a=ptime:30
<------------->
--- (18 headers 18 lines) ---
Sending to 192.168.178.1:5060 (NAT)
Using INVITE request as basis request - 049B62F3698D53D8@192.168.178.1
Found peer '192-168-178-1' for 'test' from 192.168.178.1:5060
  == Using SIP RTP CoS mark 5
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 18
Found RTP audio format 101
Found audio description format G726-32 for ID 2
Found audio description format G726-32 for ID 102
Found unknown media description format G726-40 for ID 100
Found unknown media description format G726-24 for ID 99
Found audio description format iLBC for ID 97
Found audio description format telephone-event for ID 101
Capabilities: us - 0x4 (ulaw), peer - audio=0xd0c (ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing)/text=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 192.168.178.1:7078
Looking for 666 in default (domain 192.168.178.21)
list_route: hop: <sip:test@192.168.178.1;uniq=80DE750FD696A6F05E4C556872D86>

<--- Transmitting (NAT) to 192.168.178.1:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.178.1:5060;branch=z9hG4bK8483D720AD7EC31A;received=192.168.178.1;rport=5060
From: <sip:test@192.168.178.21>;tag=810E009F9191EF51
To: <sip:666@192.168.178.21>
Call-ID: 049B62F3698D53D8@192.168.178.1
CSeq: 27 INVITE
Server: Asterisk PBX 1.8.21.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:666@192.168.178.21:5060>
Content-Length: 0


<------------>
    -- Executing [666@default:1] Dial("SIP/192-168-178-1-00000020", "SIP/666@192-168-178-20") in new stack
  == Using SIP RTP CoS mark 5
Audio is at 12542
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 192.168.178.20:5060:
INVITE sip:666@192.168.178.20 SIP/2.0
Via: SIP/2.0/UDP 192.168.178.21:5060;branch=z9hG4bK4e101628;rport
Max-Forwards: 70
From: "test" <sip:test@192.168.178.21>;tag=as673b60b8
To: <sip:666@192.168.178.20>
Contact: <sip:test@192.168.178.21:5060>
Call-ID: 139ee25452b3ce5a186b1c886547e201@192.168.178.21:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.21.0
Date: Wed, 24 Apr 2013 15:32:15 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 268

v=0
o=root 1829029832 1829029832 IN IP4 192.168.178.21
s=Asterisk PBX 1.8.21.0
c=IN IP4 192.168.178.21
t=0 0
m=audio 12542 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
    -- Called SIP/666@192-168-178-20

<--- SIP read from UDP:192.168.178.20:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.178.21:5060;branch=z9hG4bK4e101628;received=192.168.178.21;rport=5060
From: "test" <sip:test@192.168.178.21>;tag=as673b60b8
To: <sip:666@192.168.178.20>
Call-ID: 139ee25452b3ce5a186b1c886547e201@192.168.178.21:5060
CSeq: 102 INVITE
Server: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:666@192.168.178.20:5060>
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---

<--- SIP read from UDP:192.168.178.20:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.178.21:5060;branch=z9hG4bK4e101628;received=192.168.178.21;rport=5060
From: "test" <sip:test@192.168.178.21>;tag=as673b60b8
To: <sip:666@192.168.178.20>;tag=as4e2195c6
Call-ID: 139ee25452b3ce5a186b1c886547e201@192.168.178.21:5060
CSeq: 102 INVITE
Server: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:666@192.168.178.20:5060>
Content-Type: application/sdp
Content-Length: 255

v=0
o=root 1132373603 1132373603 IN IP4 192.168.178.20
s=Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
c=IN IP4 192.168.178.20
t=0 0
m=audio 17428 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (12 headers 11 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=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 192.168.178.20:17428
list_route: hop: <sip:666@192.168.178.20:5060>
set_destination: Parsing <sip:666@192.168.178.20:5060> for address/port to send to
set_destination: set destination to 192.168.178.20:5060
Transmitting (NAT) to 192.168.178.20:5060:
ACK sip:666@192.168.178.20:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.178.21:5060;branch=z9hG4bK0da431e4;rport
Max-Forwards: 70
From: "test" <sip:test@192.168.178.21>;tag=as673b60b8
To: <sip:666@192.168.178.20>;tag=as4e2195c6
Contact: <sip:test@192.168.178.21:5060>
Call-ID: 139ee25452b3ce5a186b1c886547e201@192.168.178.21:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.21.0
Content-Length: 0


---
    -- SIP/192-168-178-20-00000021 answered SIP/192-168-178-1-00000020
Audio is at 18826
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to 192.168.178.1:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.178.1:5060;branch=z9hG4bK8483D720AD7EC31A;received=192.168.178.1;rport=5060
From: <sip:test@192.168.178.21>;tag=810E009F9191EF51
To: <sip:666@192.168.178.21>;tag=as7827e31a
Call-ID: 049B62F3698D53D8@192.168.178.1
CSeq: 27 INVITE
Server: Asterisk PBX 1.8.21.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:666@192.168.178.21:5060>
Content-Type: application/sdp
Content-Length: 266

v=0
o=root 404638836 404638836 IN IP4 192.168.178.21
s=Asterisk PBX 1.8.21.0
c=IN IP4 192.168.178.21
t=0 0
m=audio 18826 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:30
a=sendrecv

<------------>

<--- SIP read from UDP:192.168.178.1:5060 --->
ACK sip:666@192.168.178.21:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.178.1:5060;branch=z9hG4bK039C57349ECDBA79
From: <sip:test@192.168.178.21>;tag=810E009F9191EF51
To: <sip:666@192.168.178.21>;tag=as7827e31a
Call-ID: 049B62F3698D53D8@192.168.178.1
CSeq: 27 ACK
Contact: <sip:test@192.168.178.1;uniq=80DE750FD696A6F05E4C556872D86>
Max-Forwards: 70
User-Agent: AVM FRITZ!Box Fon WLAN 7360 111.05.24 (Jul 6 2012)
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---

<--- SIP read from UDP:192.168.178.1:5060 --->
BYE sip:666@192.168.178.21:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.178.1:5060;rport;branch=z9hG4bK8EF5059EB9FB78A5
From: <sip:test@192.168.178.21>;tag=810E009F9191EF51
To: <sip:666@192.168.178.21>;tag=as7827e31a
Call-ID: 049B62F3698D53D8@192.168.178.1
CSeq: 28 BYE
X-RTP-Stat: CS=54;PS=784;ES=864;OS=188160;SP=0/0;SO=0;QS=-;PR=861;ER=1297;OR=206640;CR=0;SR=0;QR=-;PL=1,0;BL=1;LS=1;RB=0/0;SB=-/-;EN=PCMU;DE=PCMU;JI=98,26;DL=1,1,3;IP=192.168.178.1:7078,192.168.178.21:18826
Reason: Q.850; cause=16
Max-Forwards: 70
User-Agent: AVM FRITZ!Box Fon WLAN 7360 111.05.24 (Jul 6 2012)
Supported: 100rel,replaces
Allow-Events: telephone-event,refer
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---
Sending to 192.168.178.1:5060 (NAT)
Scheduling destruction of SIP dialog '049B62F3698D53D8@192.168.178.1' in 32000 ms (Method: BYE)

<--- Transmitting (NAT) to 192.168.178.1:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.178.1:5060;branch=z9hG4bK8EF5059EB9FB78A5;received=192.168.178.1;rport=5060
From: <sip:test@192.168.178.21>;tag=810E009F9191EF51
To: <sip:666@192.168.178.21>;tag=as7827e31a
Call-ID: 049B62F3698D53D8@192.168.178.1
CSeq: 28 BYE
Server: Asterisk PBX 1.8.21.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '139ee25452b3ce5a186b1c886547e201@192.168.178.21:5060' in 32000 ms (Method: INVITE)
set_destination: Parsing <sip:666@192.168.178.20:5060> for address/port to send to
set_destination: set destination to 192.168.178.20:5060
Reliably Transmitting (NAT) to 192.168.178.20:5060:
BYE sip:666@192.168.178.20:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.178.21:5060;branch=z9hG4bK3f6e0c54;rport
Max-Forwards: 70
From: "test" <sip:test@192.168.178.21>;tag=as673b60b8
To: <sip:666@192.168.178.20>;tag=as4e2195c6
Call-ID: 139ee25452b3ce5a186b1c886547e201@192.168.178.21:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX 1.8.21.0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---

<--- SIP read from UDP:192.168.178.20:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.178.21:5060;branch=z9hG4bK3f6e0c54;received=192.168.178.21;rport=5060
From: "test" <sip:test@192.168.178.21>;tag=as673b60b8
To: <sip:666@192.168.178.20>;tag=as4e2195c6
Call-ID: 139ee25452b3ce5a186b1c886547e201@192.168.178.21:5060
CSeq: 103 BYE
Server: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '139ee25452b3ce5a186b1c886547e201@192.168.178.21:5060' Method: INVITE
  == Spawn extension (default, 666, 1) exited non-zero on 'SIP/192-168-178-1-00000020'
Really destroying SIP dialog '049B62F3698D53D8@192.168.178.1' Method: BYE