[Freeswitch-svn] [commit] r9911 - in freeswitch/trunk/conf: . dialplan
Freeswitch SVN
brian at freeswitch.org
Thu Oct 9 01:17:16 EDT 2008
Author: brian
Date: Thu Oct 9 01:17:14 2008
New Revision: 9911
Modified:
freeswitch/trunk/conf/dialplan/default.xml
freeswitch/trunk/conf/vars.xml
Log:
tweak
Modified: freeswitch/trunk/conf/dialplan/default.xml
==============================================================================
--- freeswitch/trunk/conf/dialplan/default.xml (original)
+++ freeswitch/trunk/conf/dialplan/default.xml Thu Oct 9 01:17:14 2008
@@ -303,7 +303,7 @@
<!--<action application="set" data="conference_auto_outcall_announce=say:You have been called into an emergency conference"/>-->
<!--Add as many of these as you need, These are the people you are going to call-->
- <action application="conference_set_auto_outcall" data="sofia/gateway/asterlink.com/19184238080"/>
+ <action application="conference_set_auto_outcall" data="sofia/gateway/$${default_provider}/19184238080"/>
<action application="conference_set_auto_outcall" data="sofia/default/888 at conference.freeswitch.org"/>
<action application="conference" data="cool at default"/>
Modified: freeswitch/trunk/conf/vars.xml
==============================================================================
--- freeswitch/trunk/conf/vars.xml (original)
+++ freeswitch/trunk/conf/vars.xml Thu Oct 9 01:17:14 2008
@@ -75,6 +75,7 @@
Used by: dingaling.conf.xml
-->
<X-PRE-PROCESS cmd="set" data="bind_server_ip=auto"/>
+
<!-- external_rtp_ip
Used as the public IP address for SDP.
Can be an ip address or a string like "stun:stun.server.com"
@@ -82,6 +83,7 @@
Used by: sofia.conf.xml dingaling.conf.xml
-->
<X-PRE-PROCESS cmd="set" data="external_rtp_ip=stun:stun.freeswitch.org"/>
+
<!-- external_sip_ip
Used as the public IP address for SDP.
Can be an ip address or a string like "stun:stun.server.com"
@@ -89,10 +91,12 @@
Used by: sofia.conf.xml dingaling.conf.xml
-->
<X-PRE-PROCESS cmd="set" data="external_sip_ip=stun:stun.freeswitch.org"/>
+
<!-- unroll-loops
Used to turn on sip loopback unrolling.
-->
<X-PRE-PROCESS cmd="set" data="unroll_loops=true"/>
+
<!-- outbound_caller_id and outbound_caller_name
The caller ID telephone number we should use when calling out.
Used by: conference.conf.xml and user directory for default
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