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authorRussell Bryant <russell@russellbryant.com>2008-07-10 16:12:51 +0000
committerRussell Bryant <russell@russellbryant.com>2008-07-10 16:12:51 +0000
commitbc29f7143a74a2767b364fc6d6110e1b23be4b53 (patch)
tree9df54105afd10c06164c7b0d8721d7ae16c8e5a8 /sample.call
parentb98ac24fa1a5f48b57e2eafff4825685f749d152 (diff)
Merged revisions 129567 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ 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) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@129569 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Diffstat (limited to 'sample.call')
-rw-r--r--sample.call3
1 files changed, 2 insertions, 1 deletions
diff --git a/sample.call b/sample.call
index f00001692..eed62cfde 100644
--- a/sample.call
+++ b/sample.call
@@ -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