<div style="font-family: Arial, sans-serif; font-size: 14px;">Hi All</div><div style="font-family: Arial, sans-serif; font-size: 14px;"><br></div><div style="font-family: Arial, sans-serif; font-size: 14px;">I am using opensips as a mid_registrar for webrtc and everything is working fine except call transfers, as i understand they use refer, is there anything specific i need to change to get them to work?</div><div style="font-family: Arial, sans-serif; font-size: 14px;">if you can point me to a module id appreciate it</div><div style="font-family: Arial, sans-serif; font-size: 14px;"><br></div><div style="font-family: Arial, sans-serif; font-size: 14px;"><br></div><div style="font-family: Arial, sans-serif; font-size: 14px;"><br></div>
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