<--- SIP read from 192.168.11.125:5060 --->
INVITE sip:214057@192.168.11.124:5060;user=phone SIP/2.0
Allow:INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,REFER,SUBSCRIBE,NOTIFY,UPDATE
Via:SIP/2.0/UDP 192.168.11.125:5060;branch=z9hG4bK1771ae7036c83d2e
From:"214058" <sip:214058@192.168.11.124:5060>;tag=1987cbe6-685162
To:<sip:214057@192.168.11.124:5060;user=phone>
Call-ID:D1B9-1C8E-4668516224F9F1BAA79C-004@SipHost
CSeq:10 INVITE
Contact:<sip:214058@192.168.11.125:5060>
Expires:90
Max-Forwards:70
Remote-Party-ID:<sip:214058@192.168.11.124:5060>;party=calling;privacy=off;screen=yes
Supported:replaces
User-Agent:dlink 12-36-7924821
Content-Type:application/sdp
Content-Length:154

v=0
o=214058 1792356870 1792356870 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

<------------->
--- (15 headers 7 lines) ---
Sending to 192.168.11.125 : 5060 (no NAT)
Using INVITE request as basis request - D1B9-1C8E-4668516224F9F1BAA79C-004@SipHost

<--- Reliably Transmitting (no NAT) to 192.168.11.125:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.11.125:5060;branch=z9hG4bK1771ae7036c83d2e;received=192.168.11.125
From: "214058" <sip:214058@192.168.11.124:5060>;tag=1987cbe6-685162
To: <sip:214057@192.168.11.124:5060;user=phone>;tag=as33ee85e0
Call-ID: D1B9-1C8E-4668516224F9F1BAA79C-004@SipHost
CSeq: 10 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="795a7cb3"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'D1B9-1C8E-4668516224F9F1BAA79C-004@SipHost' in 32000 ms (Method: INVITE)
Found user '214058'
bee*CLI> 
<--- SIP read from 192.168.11.125:5060 --->
ACK sip:214057@192.168.11.124:5060;user=phone SIP/2.0
Via:SIP/2.0/UDP 192.168.11.125:5060;branch=z9hG4bK1771ae7036c83d2e
From:"214058" <sip:214058@192.168.11.124:5060>;tag=1987cbe6-685162
To:<sip:214057@192.168.11.124:5060;user=phone>;tag=as33ee85e0
Call-ID:D1B9-1C8E-4668516224F9F1BAA79C-004@SipHost
CSeq:10 ACK
Max-Forwards:70
Content-Length:0


<------------->
--- (8 headers 0 lines) ---
bee*CLI> 
<--- SIP read from 192.168.11.125:5060 --->
INVITE sip:214057@192.168.11.124:5060;user=phone SIP/2.0
Allow:INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,REFER,SUBSCRIBE,NOTIFY,UPDATE
Via:SIP/2.0/UDP 192.168.11.125:5060;branch=z9hG4bK65245ce3c5192c73
From:"214058" <sip:214058@192.168.11.124:5060>;tag=1987cbe6-685162
To:<sip:214057@192.168.11.124:5060;user=phone>
Call-ID:D1B9-1C8E-4668516224F9F1BAA79C-004@SipHost
CSeq:11 INVITE
Contact:<sip:214058@192.168.11.125:5060>
Expires:90
Max-Forwards:70
Proxy-Authorization:Digest username="214058",realm="asterisk",nonce="795a7cb3",uri="sip:214057@192.168.11.124:5060;user=phone",response="e64243b549025dd502b15818ccf083b9",algorithm=MD5
Remote-Party-ID:<sip:214058@192.168.11.124:5060>;party=calling;privacy=off;screen=yes
Supported:replaces
User-Agent:dlink 12-36-7924821
Content-Type:application/sdp
Content-Length:154

v=0
o=214058 1792356870 1792356870 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

<------------->
--- (16 headers 7 lines) ---
Sending to 192.168.11.125 : 5060 (no NAT)
Using INVITE request as basis request - D1B9-1C8E-4668516224F9F1BAA79C-004@SipHost
Found user '214058'
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
Looking for 214057 in local-phones (domain 192.168.11.124)
list_route: hop: <sip:214058@192.168.11.125:5060>

<--- Transmitting (no NAT) to 192.168.11.125:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.11.125:5060;branch=z9hG4bK65245ce3c5192c73;received=192.168.11.125
From: "214058" <sip:214058@192.168.11.124:5060>;tag=1987cbe6-685162
To: <sip:214057@192.168.11.124:5060;user=phone>
Call-ID: D1B9-1C8E-4668516224F9F1BAA79C-004@SipHost
CSeq: 11 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:214057@192.168.11.124>
Content-Length: 0


<------------>
 -- Making call to 192.168.11.126:1720 without gatekeeper.
Using 192.168.11.124 for outbound call
bee*CLI>== New H.323 Connection created.
bee*CLI>-- asterisk is calling host 192.168.11.126:1720
bee*CLI>-- Call token is ip$localhost/29414
bee*CLI>-- Call reference is 29414
bee*CLI>-- DTMF Payload is [pt=101]
Setting capabilities to 0x8 (alaw)
Capabilities in preference order is (alaw)
Allowed Codecs:
bee*CLI> Table:
   G.711-ALaw-64k <1>
   UserInput/hookflash <2>
   UserInput/RFC2833 <3>
   UserInput/dtmf <4>
 Set:LI> 
   0:LI> 
     0:> 
       G.711-ALaw-64k <1>
     1:> 
       UserInput/hookflash <2>
     2:> 
       UserInput/RFC2833 <3>
       UserInput/dtmf <4>
bee*CLI> 
bee*CLI>-- Sending SETUP message
bee*CLI>-- Transmitting RFC2833 on payload 101
bee*CLI>=-= In OnAlerting for call 29414: sessionId=0
bee*CLI>-- Ringing phone for "192.168.11.126"
bee*CLI>        - Progress Indicator: 0
bee*CLI> 
<--- Transmitting (no NAT) to 192.168.11.125:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.11.125:5060;branch=z9hG4bK65245ce3c5192c73;received=192.168.11.125
From: "214058" <sip:214058@192.168.11.124:5060>;tag=1987cbe6-685162
To: <sip:214057@192.168.11.124:5060;user=phone>;tag=as7e017701
Call-ID: D1B9-1C8E-4668516224F9F1BAA79C-004@SipHost
CSeq: 11 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:214057@192.168.11.124>
Content-Length: 0


<------------>
Using 192.168.11.124 for outbound H.245 transport
bee*CLI>ExternalRTPChannel Destroyed
bee*CLI>ExternalRTPChannel Destroyed
Peer capability is G.711-ALaw-64k <2>
Found peer capability G.711-ALaw-64k <2>, Asterisk code is 8, frame size (in ms) is 20
Peer capability is G.711-uLaw-64k <3>
Found peer capability G.711-uLaw-64k <3>, Asterisk code is 4, frame size (in ms) is 30
Peer capability is G.723.1 <4>
Found peer capability G.723.1 <4>, Asterisk code is 1, frame size (in ms) is 120
Peer capabilities = 0xd (g723|ulaw|alaw), ordered list is (alaw|ulaw|g723)
bee*CLI>-- Started logical channel: sending G.711-ALaw-64k
bee*CLI>        -- channelsOpen = 1
bee*CLI>=-= In OnConnectionEstablished for call 29414
bee*CLI>        -- Connection Established with "214057 1"
Audio is at 192.168.11.124 port 14990
Adding codec 0x8 (alaw) to SDP
bee*CLI> 
<--- Reliably 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=z9hG4bK65245ce3c5192c73;received=192.168.11.125
From: "214058" <sip:214058@192.168.11.124:5060>;tag=1987cbe6-685162
To: <sip:214057@192.168.11.124:5060;user=phone>;tag=as7e017701
Call-ID: D1B9-1C8E-4668516224F9F1BAA79C-004@SipHost
CSeq: 11 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:214057@192.168.11.124>
Content-Type: application/sdp
Content-Length: 188

v=0
o=root 20154 20154 IN IP4 192.168.11.124
s=session
c=IN IP4 192.168.11.124
t=0 0
m=audio 14990 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
bee*CLI>-- Started logical channel: receiving G.711-ALaw-64k
bee*CLI>        -- channelsOpen = 2
bee*CLI>        External RTP Session Starting
bee*CLI>        RTP channel id 1 parameters:
bee*CLI>        -- remoteIpAddress: 192.168.11.126
bee*CLI>        -- remotePort: 49586
bee*CLI>        -- ExternalIpAddress: 192.168.11.124
bee*CLI>        -- ExternalPort: 12080
bee*CLI> 
<--- SIP read from 192.168.11.125:5060 --->
ACK sip:214057@192.168.11.124 SIP/2.0
Via:SIP/2.0/UDP 192.168.11.125:5060;branch=z9hG4bK2f5376e7c032bf68
From:"214058" <sip:214058@192.168.11.124:5060>;tag=1987cbe6-685162
To:<sip:214057@192.168.11.124:5060;user=phone>;tag=as7e017701
Call-ID:D1B9-1C8E-4668516224F9F1BAA79C-004@SipHost
CSeq:11 ACK
Max-Forwards:70
Proxy-Authorization:Digest username="214058",realm="asterisk",nonce="795a7cb3",uri="sip:214057@192.168.11.124:5060;user=phone",response="e64243b549025dd502b15818ccf083b9",algorithm=MD5
User-Agent:dlink 12-36-7924821
Content-Length:0


<------------->
--- (10 headers 0 lines) ---
bee*CLI>MyH323_ExternalRTPChannel::OnReceivedAckPDU
                -- remoteIpAddress: 192.168.11.126
                -- remotePort: 49586
                External RTP Session Starting
                RTP channel id 1 parameters:
                -- remoteIpAddress: 192.168.11.126
                -- remotePort: 49586
                -- ExternalIpAddress: 192.168.11.124
                -- ExternalPort: 12080
[Jul 16 12:27:56] WARNING[20223]: rtp.c:885 ast_rtcp_read: RTCP Read too short
bee*CLI> 
<--- SIP read from 192.168.11.125:5060 --->
BYE sip:214057@192.168.11.124 SIP/2.0
Via:SIP/2.0/UDP 192.168.11.125:5060;branch=z9hG4bK13f86c7a223cbd64
From:"214058" <sip:214058@192.168.11.124:5060>;tag=1987cbe6-685162
To:<sip:214057@192.168.11.124:5060;user=phone>;tag=as7e017701
Call-ID:D1B9-1C8E-4668516224F9F1BAA79C-004@SipHost
CSeq:12 BYE
Max-Forwards:70
User-Agent:dlink 12-36-7924821
Content-Length:0


<------------->
--- (9 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=z9hG4bK13f86c7a223cbd64;received=192.168.11.125
From: "214058" <sip:214058@192.168.11.124:5060>;tag=1987cbe6-685162
To: <sip:214057@192.168.11.124:5060;user=phone>;tag=as7e017701
Call-ID: D1B9-1C8E-4668516224F9F1BAA79C-004@SipHost
CSeq: 12 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:214057@192.168.11.124>
Content-Length: 0


<------------>
bee*CLI>-- Sending RELEASE COMPLETE
        -- ClearCall: Request to clear call with token ip$localhost/29414, cause EndedByRemoteUser
bee*CLI>        channelsOpen = 1
bee*CLI>        channelsOpen = 0
bee*CLI>ExternalRTPChannel Destroyed
bee*CLI>ExternalRTPChannel Destroyed
bee*CLI>-- ClearCall: Request to clear call with token ip$localhost/29414, cause EndedByTransportFail
-- 214057 1 has cleared the call
bee*CLI>== H.323 Connection deleted.
Really destroying SIP dialog 'D1B9-1C8E-4668516224F9F1BAA79C-004@SipHost' Method: BYE