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<p class="MsoNormal">Are you going through a PBX like Asterisk? I am using rtpengine but I cannot get media packets to go from Asterisk->Kamailio(rtpengine)->softphone. I get no errors and I see rtpengine traffic, but the calls go Asterisk->softphone.<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">Thanks ALL<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal"><b>From:</b> sr-users <sr-users-bounces@lists.kamailio.org> <b>
On Behalf Of </b>Nicolas Breuer<br>
<b>Sent:</b> Monday, August 13, 2018 11:32 AM<br>
<b>To:</b> 'Kamailio (SER) - Users Mailing List' <sr-users@lists.sip-router.org><br>
<b>Subject:</b> [SR-Users] Migrate from RtpProxy to Rtpengine<o:p></o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal"><span lang="FR-BE" style="color:#002060">Hello,<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="FR-BE" style="color:#002060"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="font-size:9.0pt;font-family:"Verdana",sans-serif;color:#002060;mso-fareast-language:FR-BE">I migrated from Rtp_proxy module to Rtp_Engine.<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:9.0pt;font-family:"Verdana",sans-serif;color:#002060;mso-fareast-language:FR-BE">Everythings works fine apart the rewriting of the SDP description in Kamailio<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:9.0pt;font-family:"Verdana",sans-serif;color:#002060;mso-fareast-language:FR-BE"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="font-size:9.0pt;font-family:"Verdana",sans-serif;color:#002060;mso-fareast-language:FR-BE">subst_body('/(^s=.*)/s=Test of a Sip call.\r/')<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:9.0pt;font-family:"Verdana",sans-serif;color:#002060;mso-fareast-language:FR-BE"><o:p> </o:p></span></p>
<p class="MsoNormal">Any ideas if this is supposed to be supported?<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
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