Hi Razvan, <div><br></div><div>I got to make it work using a version downloaded from GIT. There was a mistake in the name of the file, it is working, I have removed the check from rtpproxy_stream.c</div><div><br></div><div>
Thanks for helping. </div><div><br clear="all"><div>Flavio E. Goncalves</div><div> </div><br>
<br><br><div class="gmail_quote">2011/3/11 Razvan Crainea <span dir="ltr"><<a href="mailto:razvancrainea@opensips.org">razvancrainea@opensips.org</a>></span><br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
<div bgcolor="#ffffff" text="#000000">
Hi Flavio,<br>
<br>
Can you please give me the output of the command "rtpproxy -v"?<br>
<br>
The word "session" specifies RTPProxy to search for the codec list
in the initial Update command (when OpenSIPS calls rtpproxy_offer),
so there is nothing wrong with it.<br>
<br>
This is how RTPProxy works when it receives a Play command:<br>
Assuming your initial codec list is "3,8,101", your command is
interpreted like this:<br>
it checks in the '/var/rtpproxy/prompts/' folder for the
'dmcaller.3' file (which should be a GSM encoded file). If found, it
starts to play it to the uac/uas. Otherwise it checks further for
the 'dmcaller.8' file, and so on until it finds a suitable media
file. If it doesn't, it should log an error.<br>
I guess your problem is that RTPProxy is unable to find a suitable
file to open.<br>
<br>
Regards,<br>
Razvan<div><div></div><div class="h5"><br>
<br>
On 03/11/2011 03:23 PM, Flavio Goncalves wrote:
</div></div><blockquote type="cite"><div><div></div><div class="h5">
<div>Hi, </div>
<div><br>
</div>
<div>Is there anyone with experience using rtpproxy_stream2uac
command. I'm trying to use with OpenSIPS 1.6.4 with
rtpproxy-1.2.1. I'm getting the following error:</div>
<div><br>
</div>
<div>ERROR:nathelper:rtpproxy_stream: required functionality is
not supported by the version of the RTPproxy running on the
selected node. Please upgrade the RTPproxy and try again.</div>
<div><br>
</div>
<div>1. I tried to upgrade the RTPPROXY with the on in GIT, but it
crashes as soon as I start OpenSIPS. </div>
<div><br>
</div>
<div>2. I tried to disabe the check in the rtpproxy_stream2uac, I
don't get the error but still don't work. In this case:</div>
<div><br>
</div>
<div>OpenSIPS is sending to the RTPPROXY</div>
<div><br>
</div>
<div>
<div>DBUG:handle_command: received command "23907_6 P10 <a href="mailto:f765564a-317a422b@192.168.1.175" target="_blank">f765564a-317a422b@192.168.1.175</a>
/var/rtpproxy/prompts/dmcaller session 5cc2dfbec1d32747o2;1
as1dc4b372;1"</div>
</div>
<div><br>
</div>
<div>The RTPPROXY is expecting according to the RTPPROXY protocol </div>
<div><br>
</div>
<div><span style="font-family:monospace;font-size:13px">P[args] callid play_name codecs from_tag
to_tag</span></div>
<div><span style="font-family:monospace;font-size:13px"><br>
</span></div>
<div><span style="font-family:monospace;font-size:13px">The mismatch seems to be in the word
"session" in this place rtpproxy is expecting the codecs.</span></div>
<div><span style="font-family:monospace;font-size:13px"><br>
</span></div>
<div><span style="font-family:monospace;font-size:13px">I tried everything I could. </span></div>
<div><br>
</div>
<div>Flavio E. Goncalves</div>
<div> </div>
<br>
</div></div><pre><fieldset></fieldset>
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