<div dir="ltr">Check this <a href="https://wiki.freeswitch.org/wiki/Variable_continue_on_fail">https://wiki.freeswitch.org/wiki/Variable_continue_on_fail</a></div><div class="gmail_extra"><br><br><div class="gmail_quote">On Tue, Jul 22, 2014 at 6:36 PM, Joel White <span dir="ltr"><<a href="mailto:joelewhite@gmail.com" target="_blank">joelewhite@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><div><div>I am having an issue with calls coming from another SIP server into FS. Everything works fine with the call getting to its destination. And also works if the call is ignored and goes to voicemail. However if the call is rejected by a polycom endpoint the call is dropped completely and disconnects without going to voicemail.<br>
<br></div>Here is a sample of the dialplan <br><br><br> <action application="answer"/><br> <action application="bridge" data="sofia/internal/<a href="mailto:57331@192.168.100.25" target="_blank">57331@192.168.100.25</a>"/><br>
<!--<action application="answer"/>--><br> <action application="voicemail" data="default ${domain_name} 57331"/><br> <action application="hangup"/><br>
<br><br></div>If someone could point me in the right direction<br></div>
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