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<p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Try tcpdump to see where RTP go from asterisk.<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Configure your x-lite<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Use stun server ?<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
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color:#377C17'>P</span><span style='font-size:16.0pt;font-family:"Calibri","sans-serif";
color:#377C17'> </span><span style='font-size:8.5pt;font-family:"Verdana","sans-serif";
color:#377C17'>Pensez à l'Environnement, n'imprimez ce mail que si
nécessaire.</span><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p></o:p></span></p>
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<p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>De :</span></b><span
style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>
asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <b>De la part de</b> zehra
yildiz<br>
<b>Envoyé :</b> mardi 22 décembre 2009 10:26<br>
<b>À :</b> asterisk-users@lists.digium.com<br>
<b>Objet :</b> [asterisk-users] asterisk & x-lite<o:p></o:p></span></p>
</div>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal>Hello All,<br>
<br>
I installed Asterisk 1.6.1.11 on Redhat 5.1. I use X-lite SoftPhone. The
softphone can call the other one but I can' t hear any voice. My configuration
files are below:<br>
<br>
[root@localhost asterisk]# cat sip.conf<br>
[general]<br>
canreinvite=yes<br>
<br>
[1001]<br>
username=1001<br>
password=1001<br>
type=friend<br>
context=phones<br>
host=dynamic<br>
<br>
[1002]<br>
callerid=1002<br>
username=1002<br>
password=1002<br>
type=friend<br>
context=phones<br>
host=dynamic<br>
<br>
[root@localhost asterisk]# cat extensions.conf<br>
[globals]<br>
<br>
[general]<br>
autofallthrough=yes<br>
<br>
[default]<br>
<br>
[incoming_calls]<br>
<br>
[phones]<br>
exten => _1XXX,1,NoOp()<br>
exten => _1XXX,n,Dial(SIP/${EXTEN},30)<br>
exten => _1XXX,n,Playback(the-party-you-are-calling&is-curntly-unavail)<br>
exten => _1XXX,n,Hangup()<br>
<br>
<br>
PS: My sip server and softphones are in the same network subnet. There are not
any firewall or iptables rules. I tried the "nat=yes" parameter but
no changes.<o:p></o:p></p>
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