Get a console debug log of this behavior and put it on <a href="http://pastebin.freeswitch.org">pastebin.freeswitch.org</a>, then reply to this thread with the PB URL. We'll take a peek.<br>-MC<br><br><div class="gmail_quote">
On Sun, Jun 24, 2012 at 6:17 AM, ocset <span dir="ltr"><<a href="mailto:ocset@the800group.com" target="_blank">ocset@the800group.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Hi<br>
<br>
I have a FreeSWITCH install that uses a Faktortel SIP gateway for<br>
incoming/outgoing calls. The gateway codec is g.722 and everything works<br>
fine except for redirection to voicemail. (I have hard coded the codec<br>
for the gateway since it would not work otherwise.)<br>
<br>
Here is the conf/dialplan/public/01_line1.xml file with the dialplan for<br>
incoming calls<br>
<br>
<extension name="line1"><br>
<condition field="destination_number" expression="^(5555555555)$"><br>
<action application="set" data="hangup_after_bridge=true"/><br>
<action application="set" data="continue_on_fail=true"/><br>
<action application="set" data="call_timeout=20"/><br>
<action application="bridge" data="group/sales@${domain_name}"/><br>
<action application="answer"/><br>
<action application="sleep" data="1000"/><br>
<action application="voicemail" data="default ${domain} 1000"/><br>
</condition><br>
</extension><br>
<br>
When the call is redirected to voicemail, half of the playback message<br>
is lost and the first thing the user hears is "to listen to the message<br>
press 1..."<br>
<br>
If I change the dialplan to transfer to the "1000" extension instead,<br>
allowing FreeSWITCH to redirect to voicemail automatically when ext 1001<br>
does not answer, then voicemail works correctly.<br>
<br>
Any suggestions?<br>
<br>
Thanks<br>
O.<br></blockquote></div><br>