All pastes #621514 Raw Edit

Anonymous

public text v1 · immutable
·published 2007-07-16 06:42 UTC
rendered paste body
bee*CLI>== New H.323 Connection created.
bee*CLI>--Received SETUP message
    -- Setting up Call
    --          Call token:  [ip$192.168.11.126:1798/4746]
    --          Calling party name:  [214057 1]
    --          Calling party number:  []
    --          Called party name:  [214058]
    --          Called party number:  [214058]
    --          Calling party IP:  [192.168.11.126]
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>=-= In OnAnswerCall for call 4746
bee*CLI>        - Progress Indicator: 0
bee*CLI>        - Inserting PI of 0 into ALERTING message
Audio is at 192.168.11.124 port 14516
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=z9hG4bK308ef9ed;rport
From: "Unknown" <sip:Unknown@192.168.11.124>;tag=as22ac710e
To: <sip:214058@192.168.11.125:5060>
Contact: <sip:Unknown@192.168.11.124>
Call-ID: 6fcae6215fb9037d6d6f42e80d8c5b14@192.168.11.124
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 16 Jul 2007 06:42:15 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 244

v=0
o=root 15262 15262 IN IP4 192.168.11.124
s=session
c=IN IP4 192.168.11.124
t=0 0
m=audio 14516 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=z9hG4bK308ef9ed
From:"Unknown" <sip:Unknown@192.168.11.124>;tag=as22ac710e
To:<sip:214058@192.168.11.125:5060>
Call-ID:6fcae6215fb9037d6d6f42e80d8c5b14@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=z9hG4bK308ef9ed
From:"Unknown" <sip:Unknown@192.168.11.124>;tag=as22ac710e
To:<sip:214058@192.168.11.125:5060>;tag=64cde35d-918909
Call-ID:6fcae6215fb9037d6d6f42e80d8c5b14@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>Sending alerting
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=z9hG4bK308ef9ed
From:"Unknown" <sip:Unknown@192.168.11.124>;tag=as22ac710e
To:<sip:214058@192.168.11.125:5060>;tag=64cde35d-918909
Call-ID:6fcae6215fb9037d6d6f42e80d8c5b14@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 2026106300 2026106300 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=z9hG4bK08efcfc5;rport
From: "Unknown" <sip:Unknown@192.168.11.124>;tag=as22ac710e
To: <sip:214058@192.168.11.125:5060>;tag=64cde35d-918909
Contact: <sip:Unknown@192.168.11.124>
Call-ID: 6fcae6215fb9037d6d6f42e80d8c5b14@192.168.11.124
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
bee*CLI>Answering call ip$192.168.11.126:1798/4746
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 4746
bee*CLI>        -- Connection Established with "214057 1 [192.168.11.126]"
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: 49606
bee*CLI>        -- ExternalIpAddress: 195.239.214.91
bee*CLI>        -- ExternalPort: 18734
bee*CLI>MyH323_ExternalRTPChannel::OnReceivedAckPDU
                -- remoteIpAddress: 192.168.11.126
                -- remotePort: 49606
                External RTP Session Starting
                RTP channel id 1 parameters:
                -- remoteIpAddress: 192.168.11.126
                -- remotePort: 49606
                -- ExternalIpAddress: 195.239.214.91
                -- ExternalPort: 18734
[Jul 16 10:42:23] WARNING[19810]: 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=z9hG4bKa92ce9c8d826284a
From:<sip:214058@192.168.11.125:5060>;tag=64cde35d-918909
To:"Unknown" <sip:Unknown@192.168.11.124>;tag=as22ac710e
Call-ID:6fcae6215fb9037d6d6f42e80d8c5b14@192.168.11.124
CSeq:804 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=z9hG4bKa92ce9c8d826284a;received=192.168.11.125
From: <sip:214058@192.168.11.125:5060>;tag=64cde35d-918909
To: "Unknown" <sip:Unknown@192.168.11.124>;tag=as22ac710e
Call-ID: 6fcae6215fb9037d6d6f42e80d8c5b14@192.168.11.124
CSeq: 804 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


<------------>
bee*CLI>-- Sending RELEASE COMPLETE
        -- ClearCall: Request to clear call with token ip$192.168.11.126:1798/4746, 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$192.168.11.126:1798/4746, cause EndedByTransportFail
-- 214057 1 [192.168.11.126] has cleared the call
bee*CLI>== H.323 Connection deleted.
Really destroying SIP dialog '6fcae6215fb9037d6d6f42e80d8c5b14@192.168.11.124' Method: BYE