<a href="http://pastebin.freeswitch.org/17036">http://pastebin.freeswitch.org/17036</a><div><br></div><div>I looked thru the messages, but nothing apparent came out before </div><div><br></div><div>2011-08-14 21:40:43.284111 [WARNING] rtmp.c:99 [amfnumber=2] Unhandled control packet (type=0x3)</div>
<div><br></div><div>and</div><div><br></div><div><span class="nu0">2011</span><span class="nu0">-08</span><span class="nu0">-14</span> <span class="nu0">21</span>:<span class="nu0">40</span>:<span class="nu0">43.284111</span> <span class="br0">[</span>ERR<span class="br0">]</span> rtmp_sig.c:<span class="nu0">305</span> Couldn't create call.</div>
<div><br></div><div>is there something I need to be considering on the .html/flex side? Thanks for the help.</div><div><br></div><div>--matt</div><div><br></div><div><br></div><div><br><div class="gmail_quote">On Sun, Aug 14, 2011 at 3:34 AM, Mathieu Rene <span dir="ltr"><<a href="mailto:mrene_lists@avgs.ca">mrene_lists@avgs.ca</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;"><div style="word-wrap:break-word">Switch to loglevel DEBUG (press F8), it should have a lot more clues of what's going on.<div>
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<span style="border-collapse:separate;color:rgb(0, 0, 0);font-family:Helvetica;font-size:medium;font-style:normal;font-variant:normal;font-weight:normal;letter-spacing:normal;line-height:normal;text-align:auto;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px"><div>
<span style="font-size:12px"><div>Mathieu Rene</div><div>Avant-Garde Solutions Inc</div><div>Office: <a href="tel:%2B%201%20%28514%29%20664-1044%20x100" value="+15146641044" target="_blank">+ 1 (514) 664-1044 x100</a></div>
<div>Cell: <a href="tel:%2B1%20%28514%29%20664-1044%20x200" value="+15146641044" target="_blank">+1 (514) 664-1044 x200</a></div><div><a href="mailto:mrene@avgs.ca" target="_blank">mrene@avgs.ca</a></div><div><br></div><div>
<br></div></span></div></span><br>
</div>
<br><div><div><div></div><div class="h5"><div>On 2011-08-14, at 8:10 AM, Matthew Fong wrote:</div><br></div></div><blockquote type="cite"><div><div></div><div class="h5">I am trying to get the mod_rtmp to work with the flex client, but I am receiving the following errors when I try to make a test call<div>
<br></div><div><div>2011-08-14 06:04:32.744112 [NOTICE] rtmp_sig.c:121 Sent connect reply</div>
<div>2011-08-14 06:04:38.764112 [INFO] rtmp_sig.c:136 Replied to createStream (1)</div><div>2011-08-14 06:04:38.764112 [WARNING] rtmp.c:99 [amfnumber=2] Unhandled control packet (type=0x3)</div><div>2011-08-14 06:04:38.764112 [INFO] rtmp_sig.c:136 Replied to createStream (2)</div>
<div>2011-08-14 06:04:38.764112 [NOTICE] switch_channel.c:904 New Channel rtmp/default/5000 [18578182-a702-49bc-9590-c53ec2d16b72]</div><div>2011-08-14 06:04:38.764112 [ERR] rtmp_sig.c:305 Couldn't create call.</div>
<div>
2011-08-14 06:04:38.764112 [INFO] mod_dialplan_xml.c:336 Processing <0000000000>->5000 in context public</div><div>2011-08-14 06:04:38.764112 [NOTICE] switch_core_state_machine.c:194 rtmp/default/5000 has executed the last dialplan instruction, hanging up.</div>
<div>2011-08-14 06:04:38.764112 [NOTICE] switch_core_state_machine.c:196 Hangup rtmp/default/5000 [CS_EXECUTE] [NORMAL_CLEARING]</div><div>2011-08-14 06:04:38.764112 [NOTICE] switch_core_session.c:1346 Session 1 (rtmp/default/5000) Ended</div>
<div>2011-08-14 06:04:38.764112 [NOTICE] switch_core_session.c:1348 Close Channel rtmp/default/5000 [CS_DESTROY]</div><div>2011-08-14 06:04:39.784111 [INFO] rtmp_sig.c:159 Sending audio</div><div>2011-08-14 06:04:39.784111 [WARNING] rtmp.c:99 [amfnumber=2] Unhandled control packet (type=0x3)</div>
<div>2011-08-14 06:04:39.784111 [INFO] rtmp_sig.c:274 Got publish on stream 2.</div><div>2011-08-14 06:04:39.784111 [WARNING] rtmp.c:99 [amfnumber=2] Unhandled control packet (type=0x3)</div></div><div><br></div><div>I am using a version checked-out from yesterday, Ubuntu 10.4 64-bit and flash debug version. Does anyone know what I am doing wrong? Or has anyone gotten it to work? Also I tried the FS conference call test using the hosted version on <a href="http://conference.freeswitch.org/" target="_blank">conference.freeswitch.org</a> and it seems the first few seconds of audio are distorted. Is there a way to fix this? Thanks... mod_rtmp seems very promising.</div>
<div><br></div><div>--matt</div><div><br></div></div></div>
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