[Freeswitch-users] mod_rtmp & flex client

Matthew Fong reply at matthewfong.com
Mon Aug 15 02:18:04 MSD 2011


Hi Mathieu,

I know know why I missed that. Thanks!

--matt

On Sun, Aug 14, 2011 at 2:57 PM, Mathieu Rene <mrene_lists at avgs.ca> wrote:

> The couldn't create call one should've been printed, the call was indeed
> launched, but if you look carefully, it has nowhere to go.
>
> 2011-08-14 21:40:43.284111 [NOTICE] switch_core_state_machine.c:194
> rtmp/default/5000 has executed the last dialplan instruction, hanging up.
>
> That means the dialplan didn't queue any actions for that channel.
>
> 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 11:46 PM, Matthew Fong wrote:
>
> > http://pastebin.freeswitch.org/17036
> >
> > I looked thru the messages, but nothing apparent came out before
> >
> > 2011-08-14 21:40:43.284111 [WARNING] rtmp.c:99 [amfnumber=2] Unhandled
> control packet (type=0x3)
> >
> > and
> >
> > 2011-08-14 21:40:43.284111 [ERR] rtmp_sig.c:305 Couldn't create call.
> >
> > is there something I need to be considering on the .html/flex side?
> Thanks for the help.
> >
> > --matt
> >
> >
> >
> > On Sun, Aug 14, 2011 at 3:34 AM, Mathieu Rene <mrene_lists at avgs.ca>
> wrote:
> > 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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