<html>
<head>
<meta content="text/html; charset=ISO-8859-1"
http-equiv="Content-Type">
</head>
<body bgcolor="#FFFFFF" text="#000000">
Hi,<br>
<br>
Are you using the app_rtsp under the 1.8 subdirectory? Also, could
you get an ethreal cpature of the RSTP connection?<br>
<br>
Best regards<br>
Sergio<br>
<br>
El 20/02/2012 14:49, kingman chui escribió:
<blockquote
cite="mid:1329745752.21435.YahooMailClassic@web190806.mail.sg3.yahoo.com"
type="cite">
<table border="0" cellpadding="0" cellspacing="0">
<tbody>
<tr>
<td style="font: inherit;" valign="top">
<table border="0" cellpadding="0" cellspacing="0">
<tbody>
<tr>
<td valign="top">
<div>Hi,</div>
<div> I install app_rtsp in 1.8.10.0-rc2 .</div>
<div>I cannot connect to rtsp VLC streaming server
.</div>
<div>the rtsp VLC server is working find with VLC
client and can play video in H263 format.</div>
<div>no audio .I disable audio .</div>
<div> </div>
<div>I paste the extension.conf in below </div>
<div>=======</div>
<div>exten => 2002,1,Answer()<br>
exten =>
2002,n,rtsp(rtsp://192.168.1.104:5544/)<br>
exten => 2002,n,WaitExten(5)<br>
exten => 2002,n,Hangup()</div>
<div>=======</div>
<div>I already enable the vidoe and code for the
sip.</div>
<div>[20001000]<br>
videosupport=yes<br>
allow=all</div>
<div> </div>
<div>But in the debug , the rtsp stream have not
initiate and have not play .then it end ..</div>
<div> </div>
<div>I paste part of the debug log below </div>
<div> </div>
<div>================</div>
<div>[Feb 21 05:25:26] DEBUG[11854] app_rtsp.c:
-rtsp play loop [0]<br>
[Feb 21 05:25:26] DEBUG[3636] manager.c:
Examining event:<br>
Event: Newexten^M<br>
Privilege: dialplan,all^M<br>
Channel: SIP/20001000-00000004^M<br>
Context: from-internal^M<br>
Extension: 2002^M<br>
Priority: 2^M<br>
Application: rtsp^M<br>
AppData: rtsp://192.168.1.104:5544/^M<br>
Uniqueid: 1329773126.4^M<br>
^M</div>
<div>[Feb 21 05:25:26] DEBUG[11854] app_rtsp.c:
-Receiving describe<br>
[Feb 21 05:25:26] DEBUG[11854] app_rtsp.c:
-Describe response code [200]<br>
[Feb 21 05:25:26] DEBUG[11854] app_rtsp.c:
-Receiving describe<br>
[Feb 21 05:25:26] DEBUG[11854] app_rtsp.c:
-Describe response code [2147483647]<br>
[Feb 21 05:25:26] DEBUG[11854] app_rtsp.c:
-rtsp_play end loop [0]<br>
[Feb 21 05:25:26] WARNING[11854] app_rtsp.c:
<rtsp_play<br>
[Feb 21 05:25:26] DEBUG[11854] pbx.c: Launching
'WaitExten'<br>
[Feb 21 05:25:26] VERBOSE[11854] pbx.c: --
Executing [2002@from-internal:3]
WaitExten("SIP/20001000-00000004", "5") in new
stack<br>
[Feb 21 05:25:26] DEBUG[3636] manager.c:
Examining event:<br>
Event: Newexten^M</div>
<div> </div>
<div>======</div>
<div> </div>
<div>I attach the whole debug in this email </div>
<div> </div>
<div>Please advice how to fix it .</div>
<div>I already know the app_rtsp have diff
version in asterisk 1.6 and asterisk 1.8 .</div>
<div> </div>
<div>I install the app_rtsp in asterisk 1.6 with
app_rtsp using in 1.6 before .'</div>
<div>it is working .</div>
<div>But now , I install asterisk 1.8 and use
app_rtsp with asterisk 1.8 version .</div>
<div>It is not work ..I already use the same set
up environemnt .The same as I test in asterisk
1.6 .</div>
<div> </div>
<div>Please advice why the app_rtsp using in
asterisk 1.8 version not work ..</div>
<div> </div>
<div> </div>
<div>Thank</div>
<div>Regard/chui king man</div>
</td>
</tr>
</tbody>
</table>
</td>
</tr>
</tbody>
</table>
<br>
<fieldset class="mimeAttachmentHeader"></fieldset>
<br>
<pre wrap="">--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by <a class="moz-txt-link-freetext" href="http://www.api-digital.com">http://www.api-digital.com</a> --
asterisk-video mailing list
To UNSUBSCRIBE or update options visit:
<a class="moz-txt-link-freetext" href="http://lists.digium.com/mailman/listinfo/asterisk-video">http://lists.digium.com/mailman/listinfo/asterisk-video</a></pre>
</blockquote>
<br>
</body>
</html>