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">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>