[Freeswitch-users] Zap dialplan characteristics

[Joseph L. Casale]

It seems it was permissions problems which were causing the
audio issues for me, I was attempting to run a manually
compiled instance of freeswitch with the stock init scripts.

Apparently, setting the udev rules to freeswitch/daemon as
fs runs won't work. I got it running finally tonight as a
user and group 'freeswitch' but not till after I tried the
zaptel release from the wiki's reco which didn't work until
the perms issue was discovered. I am sure I can go back to
using Digiums Dahdi package for Centos.

So last question. W/ Asterisk, I had to answer the dahdi line
so the far end didn't activate the call forward on no answer.
Would it be safe in assuming this needs to be replicated here
as well. If so, do I understand this right if I do this:

<include>
  <extension name="openzap_in">
    <condition field="source" expression="mod_openzap">
      <action application="set" data="domain_name=$${domain}"/>
      <action application="ring_ready"/>
      <action application="transfer" data="2000 XML default"/>
    </condition>
  </extension>
</include>

So that I can ring the call group for its preset time which surely
exceeds that of the call-forward from the telco?

Thanks!
jlc