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Hello Flavio,<br>
<br>
I don't know why you have to take the check out. Probably this is a
little bug. I will dig into this and let you know as soon as I solve
the problem.<br>
<br>
Regards,<br>
Razvan<br>
<br>
On 03/11/2011 04:35 PM, Flavio Goncalves wrote:
<blockquote
cite="mid:AANLkTi=UOqYVyNwx0B9i-4CRf41CTFMSxV_uNbT8U=wu@mail.gmail.com"
type="cite">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 moz-do-not-send="true"
href="mailto:razvancrainea@opensips.org">razvancrainea@opensips.org</a>></span><br>
<blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt
0.8ex; border-left: 1px solid rgb(204, 204, 204);
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 class="h5"><br>
<br>
On 03/11/2011 03:23 PM, Flavio Goncalves wrote: </div>
</div>
<blockquote type="cite">
<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 moz-do-not-send="true"
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>
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