[Freeswitch-users] rtmp help
Sherif Omran
sherifomran2000 at yahoo.com
Wed Feb 8 00:47:48 MSK 2012
Ok, now i get the following
[NO_ROUTE_DESTINATION]
could you please help?
thank you
--- On Tue, 2/7/12, rsaavedra at ecogizmos.com <rsaavedra at ecogizmos.com> wrote:
From: rsaavedra at ecogizmos.com <rsaavedra at ecogizmos.com>
Subject: Re: [Freeswitch-users] rtmp help
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Date: Tuesday, February 7, 2012, 10:18 PM
Hello,
you have to permit anonymous calls on the external profile.
Bye,
Ricardo Saavedra
> Hello all,
>
> Any body knows how to adjust the rtmp?
>
> When i try to place a call and give 1000 at serverip, I can see the call in
> fs_cli. However, it says
>
> freeswitch at internal> 2012-02-07 19:01:32.972040 [DEBUG] rtmp.c:869
> [chunk_stream=3 type=0x14 ts=12247591 stream_id=0x0] len=58
> 2012-02-07 19:01:32.972040 [DEBUG] rtmp.c:170 [amfnumber=3] Got INVOKE for
> makeCall
> 2012-02-07 19:01:32.972040 [WARNING] rtmp_sig.c:296 Unauthorized call to
> sip:1000 at serverip.com, client is not logged in.
>
> How can i make the client log in?
> I have the following in the autoconfig/rtmp.conf.xml.
>
>
> <configuration name="rtmp.conf" description="RTMP Endpoint">
> <profiles>
> <profile name="default">
> <settings>
> <param name="bind-address" value="0.0.0.0:1935" />
> <param name="context" value="sipinterface_1" />
> <param name="dialplan" value="XML" />
> <!-- If this is set to true, no unauthenticated inbound calls
> will be allowed -->
> <param name="auth-calls" value="false" />
> <!-- How much time should the clients buffer the media stream
> (miliseconds) -->
> <param name="buffer-len" value="50" />
> <!-- Sets the maximum size of outbound RTMP chunks -->
> <param name="chunksize" value="512" />
> </settings>
> </profile>
> </profiles>
> </configuration>
>
> any help is appreciated
>
> thank you
>
> regards,
> Sherif Omran
>
>
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