bee*CLI> sip set debug
SIP Debugging re-enabled
Audio is at 192.168.11.124 port 17828
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.11.125:5060:
INVITE sip:214058@192.168.11.125:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.124:5060;branch=z9hG4bK04de0bc8;rport
From: "Unknown" <sip:Unknown@192.168.11.124>;tag=as733841dd
To: <sip:214058@192.168.11.125:5060>
Contact: <sip:Unknown@192.168.11.124>
Call-ID: 448c028a2d38e6a7105416b94a3b416f@192.168.11.124
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 13 Jul 2007 13:19:00 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 244

v=0
o=root 13305 13305 IN IP4 192.168.11.124
s=session
c=IN IP4 192.168.11.124
t=0 0
m=audio 17828 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
bee*CLI> 
<--- SIP read from 192.168.11.125:5060 --->
SIP/2.0 100 Trying
Via:SIP/2.0/UDP 192.168.11.124:5060;rport;branch=z9hG4bK04de0bc8
From:"Unknown" <sip:Unknown@192.168.11.124>;tag=as733841dd
To:<sip:214058@192.168.11.125:5060>
Call-ID:448c028a2d38e6a7105416b94a3b416f@192.168.11.124
CSeq:102 INVITE
Content-Type:application/sdp
Content-Length:0


<------------->
--- (8 headers 0 lines) ---
bee*CLI> 
<--- SIP read from 192.168.11.125:5060 --->
SIP/2.0 180 Ringing
Via:SIP/2.0/UDP 192.168.11.124:5060;rport;branch=z9hG4bK04de0bc8
From:"Unknown" <sip:Unknown@192.168.11.124>;tag=as733841dd
To:<sip:214058@192.168.11.125:5060>;tag=99cdbed5-690890
Call-ID:448c028a2d38e6a7105416b94a3b416f@192.168.11.124
CSeq:102 INVITE
Contact:<sip:214058@192.168.11.125:5060>
User-Agent:dlink 12-36-7924821
Content-Length:0


<------------->
--- (9 headers 0 lines) ---
bee*CLI> 
<--- SIP read from 192.168.11.125:5060 --->
SIP/2.0 200 OK
Allow:INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,REFER,SUBSCRIBE,NOTIFY,UPDATE
Via:SIP/2.0/UDP 192.168.11.124:5060;rport;branch=z9hG4bK04de0bc8
From:"Unknown" <sip:Unknown@192.168.11.124>;tag=as733841dd
To:<sip:214058@192.168.11.125:5060>;tag=99cdbed5-690890
Call-ID:448c028a2d38e6a7105416b94a3b416f@192.168.11.124
CSeq:102 INVITE
Contact:<sip:214058@192.168.11.125:5060>
User-Agent:dlink 12-36-7924821
Content-Type:application/sdp
Content-Length:154

v=0
o=214058 1798087290 1798087290 IN IP4 192.168.11.125
s=Session SDP
c=IN IP4 192.168.11.125
t=0 0
m=audio 9000 RTP/AVP 8
a=rtpmap:8 PCMA/8000/1

<------------->
--- (11 headers 7 lines) ---
Found RTP audio format 8
Peer audio RTP is at port 192.168.11.125:9000
Found description format PCMA for ID 8
Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 192.168.11.125:9000
list_route: hop: <sip:214058@192.168.11.125:5060>
set_destination: Parsing <sip:214058@192.168.11.125:5060> for address/port to send to
set_destination: set destination to 192.168.11.125, port 5060
Transmitting (no NAT) to 192.168.11.125:5060:
ACK sip:214058@192.168.11.125:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.124:5060;branch=z9hG4bK511f9e07;rport
From: "Unknown" <sip:Unknown@192.168.11.124>;tag=as733841dd
To: <sip:214058@192.168.11.125:5060>;tag=99cdbed5-690890
Contact: <sip:Unknown@192.168.11.124>
Call-ID: 448c028a2d38e6a7105416b94a3b416f@192.168.11.124
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
[Jul 13 17:19:09] WARNING[15143]: rtp.c:885 ast_rtcp_read: RTCP Read too short
bee*CLI> 
<--- SIP read from 192.168.11.125:5060 --->
BYE sip:Unknown@192.168.11.124 SIP/2.0
Via:SIP/2.0/UDP 192.168.11.125:5060;branch=z9hG4bKde83695c4a9129cd
From:<sip:214058@192.168.11.125:5060>;tag=99cdbed5-690890
To:"Unknown" <sip:Unknown@192.168.11.124>;tag=as733841dd
Call-ID:448c028a2d38e6a7105416b94a3b416f@192.168.11.124
CSeq:32 BYE
Contact:<sip:214058@192.168.11.125:5060>
Max-Forwards:70
User-Agent:dlink 12-36-7924821
Content-Length:0


<------------->
--- (10 headers 0 lines) ---
Sending to 192.168.11.125 : 5060 (no NAT)

<--- Transmitting (no NAT) to 192.168.11.125:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.11.125:5060;branch=z9hG4bKde83695c4a9129cd;received=192.168.11.125
From: <sip:214058@192.168.11.125:5060>;tag=99cdbed5-690890
To: "Unknown" <sip:Unknown@192.168.11.124>;tag=as733841dd
Call-ID: 448c028a2d38e6a7105416b94a3b416f@192.168.11.124
CSeq: 32 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:Unknown@192.168.11.124>
Content-Length: 0


<------------>
Really destroying SIP dialog '448c028a2d38e6a7105416b94a3b416f@192.168.11.124' Method: BYE