Hello-<br>
<br>
I am attempting to use Asterisk as a voicemail server with SER doing
all of my call routing. I am running into an issue where calls
that get relayed from SER aren't getting any audio from Asterisk.
The call setup and teardown looks successful so I don't think it is a
problem with SIP. Can somebody take a look at my debugs and let
me know what they think? I've been scratching my head on this one
all morning.<br>
<br>
I've included a debug as well as SIP.conf.<br>
<br>
Thanks,<br>
Dan<br>
<br>
<br>