Hi,<br><br>I am facing audio issue in my skype for asterisk setup.<br><br><b>Flow of the call is like this.</b><br><br>e.g. <br><span style="font-weight: bold;">Skype users : </span><br>test2<br><br><span style="font-weight: bold;">Sip users:</span><br>
1001<br>1002 <--> test2<br><br><span style="font-weight: bold;">This both sip users 1001 and 1002 are register in same asterisk. And also test2 skype user is register in same asterisk.<br><br>Now 1001 is dialing skype user test2 (skypeout). So, test2 is getting call. But as test2 skype user is register in our asterisk, our asterisk is getting that call (skypein). And test2 is mapped with 1002 user. So when test2 user call comes to asterisk our asterisk is dialing SIP/1002. And 1002 is getting calls. But when 1001 and 1002 user is connecting they are not getting audio. But this is working fine for only skypout and skypein. But when call come back to asterisk audio issue is coming.<br>
</span><br>I have checked rtp debug, But getting proper packages in rtp debug.<br><br>I am attaching image of call flow.<br><img title="?ui=2&view=att&th=1247ba299ad2be9d&attid=0.1&disp=attd&realattid=ii_1247ba299ad2be9d&zw" alt="?ui=2&view=att&th=1247ba299ad2be9d&attid=0.1&disp=attd&realattid=ii_1247ba299ad2be9d&zw" src="cid:ii_1247ba299ad2be9d"><br>
<br>Please help me to fix the issue.<br><br>-- <br><br>Thanks,<br>Samir Doshi<br>