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Merged revisions 129567 via svnmerge from 

https://origsvn.digium.com/svn/asterisk/branches/1.4

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r129567 | russell | 2008-07-10 11:03:59 -0500 (Thu, 10 Jul 2008) | 3 lines

Note that pbx_spool.so is the module used for call files
(inspired by a question in #asterisk)

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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@129569 f38db490-d61c-443f-a65b-d21fe96a405b
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russell 2008-07-10 16:12:51 +00:00
parent fe2ac7bf3f
commit 148b74212e
1 changed files with 2 additions and 1 deletions
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3
sample.call
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@ -1,6 +1,7 @@
#
# This is a sample file that can be dumped in /var/spool/asterisk/outgoing
# to generate a call.
# to generate a call. For Asterisk to read call files, you must have the
# pbx_spool.so module loaded.
#
# Comments are indicated by a '#' character that begins a line, or follows
# a space or tab character. To be consistent with the configuration files