<div style="color:#000; background-color:#fff; font-family:times new roman, new york, times, serif;font-size:12pt"><div><span><br></span></div><div>&nbsp;</div><div>Thanks Giovanni for you respose, you understandme <span id="result_box" class="short_text" lang="en"><span class="hps">perfectly.</span></span></div><div><span id="result_box" class="short_text" lang="en"><span class="hps">A frew aditional information:</span></span></div><div><span id="result_box" class="short_text" lang="en"><span class="hps">My SIP (DID) provider does not requiere a password,&nbsp; </span></span><span id="result_box" class="" lang="en"><span class="hps">and</span> <span class="hps">the configuration</span> <span class="hps">does not require</span> <span class="hps">any</span> <span class="hps">specific</span> <span class="hps">parameter. I put </span></span>SIP/<a href="/user/SendEmail.jtp?type=node&node=7036703&i=0" target="_top" rel="nofollow" link="external">[hidden email]</a>:5080 at provider mapping.<br></div><div>In Freeswitch:<br></div><div><span
 id="result_box" class="" lang="en"><span class="hps">I made change only in two files:</span></span></div><div><span id="result_box" class="" lang="en"><span class="hps">acl.conf.xml - for include in domains list the providers IP'S</span></span></div><div><span id="result_box" class="" lang="en"><span class="hps">public.xml - for include this:</span></span></div><div><span id="result_box" class="" lang="en"><span class="hps">&lt;extension name="skype-call"&gt;<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &lt;condition field="destination_number" expression="^5001$"&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &lt;action application="set" data="skype_get_inband_dtmf=true"/&gt;<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &lt;action application="bridge" data="skypopen/RR/skype22"/&gt;&nbsp; &lt;--- skype user<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &lt;/condition&gt;<br>&nbsp;&nbsp;&nbsp; &lt;/extension&gt;</span></span></div><div>skypopen.conf.xml, dont
 change anything at this moment the file are same as source.<br></div><div><br></div><div id="gt-src-c" class="g-unit"><div id="gt-src-p"> </div> </div>  <div style="font-family: times new roman, new york, times, serif; font-size: 12pt;"> <div style="font-family: times new roman, new york, times, serif; font-size: 12pt;"> <font size="2" face="Arial"> <hr size="1">  <b><span style="font-weight:bold;">De:</span></b> Giovanni Maruzzelli-2 [via freeswitch-users] &lt;<a href="/user/SendEmail.jtp?type=node&node=7036703&i=1" target="_top" rel="nofollow" link="external">[hidden email]</a>&gt;<br> <b><span style="font-weight: bold;">Para:</span></b> JoZu &lt;<a href="/user/SendEmail.jtp?type=node&node=7036703&i=2" target="_top" rel="nofollow" link="external">[hidden email]</a>&gt; <br> <b><span style="font-weight: bold;">Enviado:</span></b> domingo 27 de noviembre de 2011 17:31<br> <b><span style="font-weight: bold;">Asunto:</span></b> Re: Skypopen audio problems<br> </font> <br><meta http-equiv="x-dns-prefetch-control" content="off"><div id="yiv159725393">

        No, skypopen does not need a soundcard.
<br><br>&gt;From your log nothing strange in skypopen.
<br><br>Let me recap if I have understood:
<br><br>1) you have a SIP DID
<br>2) this SIP DID is going to an extension that bridges to a skypopen
<br>outbound call to SKYPE_22
<br>3) the SIP incoming call is correctly bridged to SKYPE_22
<br>4) DID caller can hear SKYPE_22 talking
<br>5) SKYPE_22 do not hear DID caller talking. This is the problem.
<br><br>I have understood correctly?
<br><br>Can you post the skypopen.conf.xml and the dialplan snippet that do the bridge?
<br><br>-giovanni
<br><br>On Sun, Nov 27, 2011 at 2:00 PM, asdfa asdfasdf &lt;<a rel="nofollow" target="_top" link="external">[hidden email]</a>&gt; wrote:
<div class="yiv159725393shrinkable-quote"><div class='shrinkable-quote'><br>&gt; Hi, i have a little problem i have FS installed, skypeopen_oss client and
<br>&gt; skypeopen.ko oss sound driver installed.
<br>&gt; I configure in public a DID number bridge to skypopen/RR/skype22, the call
<br>&gt; going from DID to SKYPE1 and call SKYPE22, here is the LOG:
<br>&gt; <a rel="nofollow" target="_blank" href="http://pastebin.freeswitch.org/17877" link="external">http://pastebin.freeswitch.org/17877</a><br>&gt; All work OK, but the sound from DID caller to Skype22 dont reach, no sound,
<br>&gt; Skype22 to DID caller works ok. Questions:
<br>&gt; Skype_oss and the skypeopen.ko require a sound card to work?
<br>&gt; What is the problem? Codecs?
<br>&gt; Please helpme, thanks and sorry my english.
<br>&gt;
<br>&gt;
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</div></div><br><br>-- 
<br>Sincerely,
<br><br>Giovanni Maruzzelli
<br>Cell : +39-347-2665618
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