[Asterisk-video] app_rtsp

Sergio Garcia Murillo sergio.garcia at fontventa.com
Fri Aug 22 03:05:18 CDT 2008


Hi Rene,

How do u originate the call to the 3g?
Also, could you enable the debug trace and check for the following trace ?

ast_log(LOG_DEBUG,"-rtsp_play end loop [%d]\n",res);


Best regards
Sergio

----- Original Message -----
From: Rene van Weert [mailto:rvweert at gmail.com]
To: asterisk-video at lists.digium.com
Sent: Fri, 8 Aug 2008 14:57:01 +0200
Subject: [Asterisk-video] app_rtsp

Hi All,

I'm having some trouble with app_rtsp. Maybe someone can help me out..
When entering dtmf digits asterisk doesn't jump to the entered extension but
just goed to the next priority.
Further when I orginate a call to a 3g handset i get the following error
when starting rtsp playback:
[Aug  8 14:55:51] WARNING[6411]: channel.c:2736 set_format: Unable to find a
codec translation path from slin to unknown
When I do an inbound call however the rtsp stream works perfectly, with
audio and video...

Any clues?

Thanks in advance.

Cheers,

Rene van Weert



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