[Freeswitch-users] mod_rtmp & flex client

[Mathieu Rene]

Switch to loglevel DEBUG (press F8), it should have a lot more clues of what's going on.
 
Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 2011-08-14, at 8:10 AM, Matthew Fong wrote:

> 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
> 
> 2011-08-14 06:04:32.744112 [NOTICE] rtmp_sig.c:121 Sent connect reply
> 2011-08-14 06:04:38.764112 [INFO] rtmp_sig.c:136 Replied to createStream (1)
> 2011-08-14 06:04:38.764112 [WARNING] rtmp.c:99 [amfnumber=2] Unhandled control packet (type=0x3)
> 2011-08-14 06:04:38.764112 [INFO] rtmp_sig.c:136 Replied to createStream (2)
> 2011-08-14 06:04:38.764112 [NOTICE] switch_channel.c:904 New Channel rtmp/default/5000 [18578182-a702-49bc-9590-c53ec2d16b72]
> 2011-08-14 06:04:38.764112 [ERR] rtmp_sig.c:305 Couldn't create call.
> 2011-08-14 06:04:38.764112 [INFO] mod_dialplan_xml.c:336 Processing  <0000000000>->5000 in context public
> 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.
> 2011-08-14 06:04:38.764112 [NOTICE] switch_core_state_machine.c:196 Hangup rtmp/default/5000 [CS_EXECUTE] [NORMAL_CLEARING]
> 2011-08-14 06:04:38.764112 [NOTICE] switch_core_session.c:1346 Session 1 (rtmp/default/5000) Ended
> 2011-08-14 06:04:38.764112 [NOTICE] switch_core_session.c:1348 Close Channel rtmp/default/5000 [CS_DESTROY]
> 2011-08-14 06:04:39.784111 [INFO] rtmp_sig.c:159 Sending audio
> 2011-08-14 06:04:39.784111 [WARNING] rtmp.c:99 [amfnumber=2] Unhandled control packet (type=0x3)
> 2011-08-14 06:04:39.784111 [INFO] rtmp_sig.c:274 Got publish on stream 2.
> 2011-08-14 06:04:39.784111 [WARNING] rtmp.c:99 [amfnumber=2] Unhandled control packet (type=0x3)
> 
> 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 conference.freeswitch.org and it seems the first few seconds of audio are distorted. Is there a way to fix this? Thanks... mod_rtmp seems very promising.
> 
> --matt
> 
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