[Freeswitch-svn] [commit] r3635 - in freeswitch/branches/knhor/trunk: . build conf libs/libdingaling libs/libdingaling/src libs/libresample/src src/mod/applications/mod_conference src/mod/endpoints/mod_dingaling src/mod/endpoints/mod_portaudio

Freeswitch SVN knhor at freeswitch.org
Wed Dec 13 13:04:49 EST 2006


Author: knhor
Date: Wed Dec 13 13:04:49 2006
New Revision: 3635

Removed:
   freeswitch/branches/knhor/trunk/libs/libresample/src/config.h
Modified:
   freeswitch/branches/knhor/trunk/   (props changed)
   freeswitch/branches/knhor/trunk/build/buildlib.sh
   freeswitch/branches/knhor/trunk/conf/freeswitch.xml
   freeswitch/branches/knhor/trunk/libs/libdingaling/.update
   freeswitch/branches/knhor/trunk/libs/libdingaling/src/libdingaling.c
   freeswitch/branches/knhor/trunk/libs/libdingaling/src/libdingaling.h
   freeswitch/branches/knhor/trunk/libs/libresample/src/   (props changed)
   freeswitch/branches/knhor/trunk/src/mod/applications/mod_conference/mod_conference.c
   freeswitch/branches/knhor/trunk/src/mod/endpoints/mod_dingaling/mod_dingaling.c
   freeswitch/branches/knhor/trunk/src/mod/endpoints/mod_portaudio/mod_portaudio.c

Log:
Merged revisions 3599-3634 via svnmerge from trunk


Modified: freeswitch/branches/knhor/trunk/build/buildlib.sh
==============================================================================
--- freeswitch/branches/knhor/trunk/build/buildlib.sh	(original)
+++ freeswitch/branches/knhor/trunk/build/buildlib.sh	Wed Dec 13 13:04:49 2006
@@ -20,6 +20,11 @@
     fi
 fi
 
+GZCAT=`which gzcat 2>/dev/null`
+if [ -z "$GZCAT" ] ; then
+    GZCAT=zcat
+fi
+
 install=
 base=http://svn.freeswitch.org/downloads/libs
 
@@ -52,15 +57,18 @@
 	fi
     fi
     if [ ! -d $uncompressed ] ; then
-	tar -zxvf $tar
+	$GZCAT $tar | tar xf -
     fi
 fi
-
-if [ ! -f $root/.nothanks ] && [ $uncompressed/.complete -ot $uncompressed ] ; then 
+if [ -f $uncompressed/.complete ] ; then 
+if [ ! -n "$(find $uncompressed/.complete -prune -newer $uncompressed)" ]; then
+if [ ! -f $root/.nothanks ] ; then 
     echo remove stale .complete
     rm $uncompressed/.complete
     sh -c "cd $uncompressed && $MAKE clean distclean"
 fi
+fi
+fi
 
 if [ -f $uncompressed/.complete ] ; then
     echo $uncompressed already installed
@@ -88,6 +96,8 @@
 fi
 
 if [ $? = 0 ] ; then
+    touch .complete
+    sleep 1
     touch .complete
 else 
     echo ERROR

Modified: freeswitch/branches/knhor/trunk/conf/freeswitch.xml
==============================================================================
--- freeswitch/branches/knhor/trunk/conf/freeswitch.xml	(original)
+++ freeswitch/branches/knhor/trunk/conf/freeswitch.xml	Wed Dec 13 13:04:49 2006
@@ -5,8 +5,8 @@
     
     <configuration name="switch.conf" description="Modules">
       <settings>
-        <!--Most channels to allow at once -->
-        <param name="max-sessions" value="1000"/>
+	<!--Most channels to allow at once -->
+	<param name="max-sessions" value="1000"/>
       </settings>
       <!--Any variables defined here will be available in every channel, in the dialplan etc -->
       <variables>
@@ -18,70 +18,70 @@
 
     <configuration name="modules.conf" description="Modules">
       <modules>
-        <!-- Loggers (I'd load these first) -->
-        <load module="mod_console"/>
-        <!-- <load module="mod_syslog"/> -->
+	<!-- Loggers (I'd load these first) -->
+	<load module="mod_console"/>
+	<!-- <load module="mod_syslog"/> -->
 
-        <!-- Multi-Faceted -->
+	<!-- Multi-Faceted -->
 	<!-- mod_enum is a dialplan interface, an application interface and an api command interface -->
-        <load module="mod_enum"/>
+	<load module="mod_enum"/>
 
-        <!-- XML Interfaces -->
-        <!-- <load module="mod_xml_rpc"/> -->
-        <!-- <load module="mod_xml_curl"/> -->
+	<!-- XML Interfaces -->
+	<!-- <load module="mod_xml_rpc"/> -->
+	<!-- <load module="mod_xml_curl"/> -->
 
-        <!-- Event Handlers -->
-        <!-- <load module="mod_cdr"/> -->
-        <!-- <load module="mod_event_multicast"/> -->
-        <!-- <load module="mod_event_socket"/> -->
-        <!-- <load module="mod_xmpp_event"/> -->
-        <!-- <load module="mod_zeroconf"/> -->
+	<!-- Event Handlers -->
+	<!-- <load module="mod_cdr"/> -->
+	<!-- <load module="mod_event_multicast"/> -->
+	<!-- <load module="mod_event_socket"/> -->
+	<!-- <load module="mod_xmpp_event"/> -->
+	<!-- <load module="mod_zeroconf"/> -->
 
-        <!-- Directory Interfaces -->
-        <!-- <load module="mod_ldap"/> -->
+	<!-- Directory Interfaces -->
+	<!-- <load module="mod_ldap"/> -->
 
-        <!-- Endpoints -->
-        <!-- <load module="mod_dingaling"/> -->
-        <!--<load module="mod_iax"/>-->
-        <load module="mod_portaudio"/>
-        <load module="mod_sofia"/>
-        <!-- <load module="mod_wanpipe"/> -->
-        <!-- <load module="mod_woomera"/> -->
+	<!-- Endpoints -->
+	<!-- <load module="mod_dingaling"/> -->
+	<!--<load module="mod_iax"/>-->
+	<load module="mod_portaudio"/>
+	<load module="mod_sofia"/>
+	<!-- <load module="mod_wanpipe"/> -->
+	<!-- <load module="mod_woomera"/> -->
 
-        <!-- Applications -->
-        <load module="mod_bridgecall"/>
-        <load module="mod_commands"/>
-        <load module="mod_conference"/>
-        <load module="mod_dptools"/>
-        <load module="mod_echo"/>
-        <!--<load module="mod_park"/>-->
-        <load module="mod_playback"/>
+	<!-- Applications -->
+	<load module="mod_bridgecall"/>
+	<load module="mod_commands"/>
+	<load module="mod_conference"/>
+	<load module="mod_dptools"/>
+	<load module="mod_echo"/>
+	<!--<load module="mod_park"/>-->
+	<load module="mod_playback"/>
 
-        <!-- Dialplan Interfaces -->
-        <!-- <load module="mod_dialplan_directory"/> -->
-        <load module="mod_dialplan_xml"/>
+	<!-- Dialplan Interfaces -->
+	<!-- <load module="mod_dialplan_directory"/> -->
+	<load module="mod_dialplan_xml"/>
 
-        <!-- Codec Interfaces -->
-        <load module="mod_g711"/>
-        <load module="mod_gsm"/>
-        <!-- <load module="mod_ilbc"/> -->
-        <load module="mod_l16"/>
-        <!-- <load module="mod_speex"/> -->
+	<!-- Codec Interfaces -->
+	<load module="mod_g711"/>
+	<load module="mod_gsm"/>
+	<!-- <load module="mod_ilbc"/> -->
+	<load module="mod_l16"/>
+	<!-- <load module="mod_speex"/> -->
 
-        <!-- File Format Interfaces -->
-        <load module="mod_sndfile"/>
-        <load module="mod_native_file"/>
+	<!-- File Format Interfaces -->
+	<load module="mod_sndfile"/>
+	<load module="mod_native_file"/>
 
-        <!-- Timers -->
-        <load module="mod_softtimer"/>
+	<!-- Timers -->
+	<load module="mod_softtimer"/>
 
-        <!-- Languages -->
-        <!-- <load module="mod_spidermonkey"/> -->
-        <!-- <load module="mod_perl"/> -->
+	<!-- Languages -->
+	<!-- <load module="mod_spidermonkey"/> -->
+	<!-- <load module="mod_perl"/> -->
 
-        <!-- ASR /TTS -->
-        <!-- <load module="mod_cepstral"/> -->
-        <!-- <load module="mod_rss"/> -->
+	<!-- ASR /TTS -->
+	<!-- <load module="mod_cepstral"/> -->
+	<!-- <load module="mod_rss"/> -->
       </modules>
     </configuration>
 
@@ -95,29 +95,29 @@
 
     <configuration name="event_multicast.conf" description="Multicast Event">
       <settings>
-        <param name="address" value="225.1.1.1"/>
-        <param name="port" value="4242"/>
-        <param name="bindings" value="all"/>
+	<param name="address" value="225.1.1.1"/>
+	<param name="port" value="4242"/>
+	<param name="bindings" value="all"/>
       </settings>
     </configuration>
 
     <configuration name="event_socket.conf" description="Socket Client">
       <settings>
-        <param name="listen-ip" value="127.0.0.1"/>
-        <param name="listen-port" value="8021"/>
-        <param name="password" value="ClueCon"/>
+	<param name="listen-ip" value="127.0.0.1"/>
+	<param name="listen-port" value="8021"/>
+	<param name="password" value="ClueCon"/>
       </settings>
     </configuration>
 
     <configuration name="iax.conf" description="IAX Configuration">
       <settings>
-        <param name="debug" value="0"/>
-        <!-- <param name="ip" value="1.2.3.4"> -->
-        <param name="port" value="4569"/>
-        <param name="dialplan" value="XML"/>
-        <param name="codec-prefs" value="PCMU at 20i,PCMA,speex,L16"/>
-        <param name="codec-master" value="us"/>
-        <param name="codec-rates" value="8"/>
+	<param name="debug" value="0"/>
+	<!-- <param name="ip" value="1.2.3.4"> -->
+	<param name="port" value="4569"/>
+	<param name="dialplan" value="XML"/>
+	<param name="codec-prefs" value="PCMU at 20i,PCMA,speex,L16"/>
+	<param name="codec-master" value="us"/>
+	<param name="codec-rates" value="8"/>
       </settings>
     </configuration>
 
@@ -125,57 +125,57 @@
       <!-- pick a file name, a function name or 'all' -->
       <!-- map as many as you need for specific debugging -->
       <mappings>
-        <!-- <param name="log_event" value="DEBUG"/> -->
-        <param name="all" value="DEBUG"/>
+	<!-- <param name="log_event" value="DEBUG"/> -->
+	<param name="all" value="DEBUG"/>
       </mappings>
     </configuration>
 
     <configuration name="sofia.conf" description="sofia Endpoint">
       <profiles>
-        <profile name="mydomain1.com">
-          <registrations>
-            <!-- <registration name="asterlink">
-              <param name="register-scheme" value="Digest"/>
-              <param name="register-realm" value=""/>
-              <param name="register-username" value="1001"/>
-              <param name="register-password" value="nhy65tgb"/>
-              <param name="register-from" value="sip:1001 at 208.64.200.40"/>
-              <param name="register-to" value="sip:1001 at conference.freeswitch.org"/>
-              <param name="register-proxy" value="sip:conference.freeswitch.org:5060"/>
-              <param name="register-frequency" value="20"/>
-            </registration> -->
-          </registrations>
-          <settings>
-            <param name="debug" value="1"/>
-            <param name="rfc2833-pt" value="101"/>
-            <param name="sip-port" value="5060"/>
-            <param name="dialplan" value="XML"/>
-            <param name="dtmf-duration" value="100"/>
-            <param name="codec-prefs" value="PCMU at 20i"/>
-            <param name="codec-ms" value="20"/>
-            <param name="use-rtp-timer" value="true"/>
-            <param name="rtp-timer-name" value="soft"/>
-            <param name="rtp-ip" value="192.168.1.20"/>
-            <param name="sip-ip" value="mydomain1.com"/>
+	<profile name="mydomain1.com">
+	  <registrations>
+	    <!-- <registration name="asterlink">
+		 <param name="register-scheme" value="Digest"/>
+		 <param name="register-realm" value=""/>
+		 <param name="register-username" value="1001"/>
+		 <param name="register-password" value="nhy65tgb"/>
+		 <param name="register-from" value="sip:1001 at 208.64.200.40"/>
+		 <param name="register-to" value="sip:1001 at conference.freeswitch.org"/>
+		 <param name="register-proxy" value="sip:conference.freeswitch.org:5060"/>
+		 <param name="register-frequency" value="20"/>
+		 </registration> -->
+	  </registrations>
+	  <settings>
+	    <param name="debug" value="1"/>
+	    <param name="rfc2833-pt" value="101"/>
+	    <param name="sip-port" value="5060"/>
+	    <param name="dialplan" value="XML"/>
+	    <param name="dtmf-duration" value="100"/>
+	    <param name="codec-prefs" value="PCMU at 20i"/>
+	    <param name="codec-ms" value="20"/>
+	    <param name="use-rtp-timer" value="true"/>
+	    <param name="rtp-timer-name" value="soft"/>
+	    <param name="rtp-ip" value="192.168.1.20"/>
+	    <param name="sip-ip" value="mydomain1.com"/>
 
-            <!-- this lets anything register -->
-            <!--  comment the next line and uncomment one or both of the other 2 lines for call authentication -->
-            <param name="accept-blind-reg" value="true"/>
+	    <!-- this lets anything register -->
+	    <!--  comment the next line and uncomment one or both of the other 2 lines for call authentication -->
+	    <param name="accept-blind-reg" value="true"/>
 
-            <!--<param name="auth-calls" value="true"/>-->
-            <!-- on authed calls, authenticate *all* the packets not just invite -->
-            <!--<param name="auth-all-packets" value="true"/>-->
+	    <!--<param name="auth-calls" value="true"/>-->
+	    <!-- on authed calls, authenticate *all* the packets not just invite -->
+	    <!--<param name="auth-all-packets" value="true"/>-->
 
-            <!-- optional ; -->
-            <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>-->
-            <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> -->
-            <!-- VAD choose one (out is a good choice); -->
-            <!-- <param name="vad" value="in"/> -->
-            <!-- <param name="vad" value="out"/> -->
-            <!-- <param name="vad" value="both"/> -->
-            <!--<param name="alias" value="sip:10.0.1.251:5555"/>-->
-          </settings>
-        </profile>
+	    <!-- optional ; -->
+	    <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>-->
+	    <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> -->
+	    <!-- VAD choose one (out is a good choice); -->
+	    <!-- <param name="vad" value="in"/> -->
+	    <!-- <param name="vad" value="out"/> -->
+	    <!-- <param name="vad" value="both"/> -->
+	    <!--<param name="alias" value="sip:10.0.1.251:5555"/>-->
+	  </settings>
+	</profile>
       </profiles>
     </configuration>
 
@@ -190,147 +190,147 @@
       <!-- info    - informational message  -->
       <!-- debug   - debug-level message -->
       <settings>
-        <param name="ident" value="freeswitch"/>
-        <param name="facility" value="user"/>
-        <param name="format" value="${time} - ${message}"/>
-        <param name="level" value="debug,info,warning-alert"/>
+	<param name="ident" value="freeswitch"/>
+	<param name="facility" value="user"/>
+	<param name="format" value="${time} - ${message}"/>
+	<param name="level" value="debug,info,warning-alert"/>
       </settings>
     </configuration>
 
     <configuration name="woomera.conf" description="Woomera Endpoint">
       <settings>
-        <param name="debug" value="0"/>
+	<param name="debug" value="0"/>
       </settings>
       <interface>
-        <param name="host" value="localhost"/>
-        <param name="port" value="42420"/>
-        <param name="audio-ip" value="127.0.0.1"/>
-        <param name="dialplan" value="XML"/>
+	<param name="host" value="localhost"/>
+	<param name="port" value="42420"/>
+	<param name="audio-ip" value="127.0.0.1"/>
+	<param name="dialplan" value="XML"/>
       </interface>
     </configuration>
 
     <configuration name="wanpipe.conf" description="Sangoma Wanpipe Endpoint">
       <settings>
-        <param name="debug" value="1"/>
-        <param name="dialplan" value="XML"/>
-        <param name="mtu" value="320"/>
-        <param name="dtmf-on" value="800"/>
-        <param name="dtmf-off" value="100"/>
-        <param name="supress-dtmf-tone" value="yes"/>
+	<param name="debug" value="1"/>
+	<param name="dialplan" value="XML"/>
+	<param name="mtu" value="320"/>
+	<param name="dtmf-on" value="800"/>
+	<param name="dtmf-off" value="100"/>
+	<param name="supress-dtmf-tone" value="yes"/>
       </settings>
       <span>
-        <param name="span" value="1"/>
-        <param name="node" value="cpe"/>
-        <!-- <param name="switch" value="ni2"/> -->
-        <param name="switch" value="dms100"/>
-        <!-- <param name="switch" value="lucent5e"/> -->
-        <!-- <param name="switch" value="att4ess"/> -->
-        <!-- <param name="switch" value="euroisdn"/> -->
-        <!-- <param name="switch" value="gr303eoc"/> -->
-        <!-- <param name="switch" value="gr303tmc"/> -->
-        <param name="dp" value="national"/>
-        <!-- <param name="dp" value="international"/> -->
-        <!-- <param name="dp" value="local"/> -->
-        <!-- <param name="dp" value="private"/> -->
-        <!-- <param name="dp" value="unknown"/> -->
-        <param name="l1" value="ulaw"/>
-        <!-- <param name="l1" value="alaw"/> -->
-        <param name="bchan" value="1-23"/>
-        <param name="dchan" value="24"/>
-        <param name="dialplan" value="XML"/>
+	<param name="span" value="1"/>
+	<param name="node" value="cpe"/>
+	<!-- <param name="switch" value="ni2"/> -->
+	<param name="switch" value="dms100"/>
+	<!-- <param name="switch" value="lucent5e"/> -->
+	<!-- <param name="switch" value="att4ess"/> -->
+	<!-- <param name="switch" value="euroisdn"/> -->
+	<!-- <param name="switch" value="gr303eoc"/> -->
+	<!-- <param name="switch" value="gr303tmc"/> -->
+	<param name="dp" value="national"/>
+	<!-- <param name="dp" value="international"/> -->
+	<!-- <param name="dp" value="local"/> -->
+	<!-- <param name="dp" value="private"/> -->
+	<!-- <param name="dp" value="unknown"/> -->
+	<param name="l1" value="ulaw"/>
+	<!-- <param name="l1" value="alaw"/> -->
+	<param name="bchan" value="1-23"/>
+	<param name="dchan" value="24"/>
+	<param name="dialplan" value="XML"/>
       </span>
     </configuration>
 
     <configuration name="portaudio.conf" description="Soundcard Endpoint">
       <settings>
-        <param name="debug" value="2"/>
-        <param name="dialplan" value="XML"/>
+	<param name="debug" value="2"/>
+	<param name="dialplan" value="XML"/>
 
-        <!-- partial string match on something in the name or the device # -->
-        <param name="indev" value="USB"/>
-        <param name="outdev" value="USB"/>
+	<!-- partial string match on something in the name or the device # -->
+	<param name="indev" value="USB"/>
+	<param name="outdev" value="USB"/>
 
-        <param name="cid-name" value="FreeSwitch"/>
-        <param name="cid-num" value="5555551212"/>
+	<param name="cid-name" value="FreeSwitch"/>
+	<param name="cid-num" value="5555551212"/>
       </settings>
     </configuration>
 
     <configuration name="zeroconf.conf" description="Zeroconf Event Handler">
       <settings>
-        <param name="publish" value="yes"/>
-        <param name="browse" value="_sip._udp"/>
+	<param name="publish" value="yes"/>
+	<param name="browse" value="_sip._udp"/>
       </settings>
     </configuration>
 
     <configuration name="xmpp_event.conf" description="XMPP Event Handler">
       <settings>
-        <param name="#debug" value="1"/>
-        <param name="jid" value="freeswitch at my.jabber.com/me"/>
-        <param name="passwd" value="mypass"/>
-        <param name="target-jid" value="freeswitch at reader.org/him"/>
+	<param name="#debug" value="1"/>
+	<param name="jid" value="freeswitch at my.jabber.com/me"/>
+	<param name="passwd" value="mypass"/>
+	<param name="target-jid" value="freeswitch at reader.org/him"/>
       </settings>
     </configuration>
 
     <configuration name="dialplan_directory.conf" description="Dialplan Directory">
       <settings>
-        <param name="directory-name" value="ldap"/>
-        <param name="host" value="ldap.freeswitch.org"/>
-        <param name="dn" value="cn=Manager,dc=freeswitch,dc=org"/>
-        <param name="pass" value="test"/>
-        <param name="base" value="dc=freeswitch,dc=org"/>
+	<param name="directory-name" value="ldap"/>
+	<param name="host" value="ldap.freeswitch.org"/>
+	<param name="dn" value="cn=Manager,dc=freeswitch,dc=org"/>
+	<param name="pass" value="test"/>
+	<param name="base" value="dc=freeswitch,dc=org"/>
       </settings>
     </configuration>
 
     <configuration name="dingaling.conf" description="XMPP Jingle Endpoint">
       <settings>
-        <param name="debug" value="0"/>
-        <param name="codec-prefs" value="PCMU"/>
+	<param name="debug" value="0"/>
+	<param name="codec-prefs" value="PCMU"/>
       </settings>
 
       <!-- *NOTE* change <x-profile></x-profile> to <profile></profile> to enable -->
 
       <!-- Client Profile (Original mode) -->
       <x-profile type="client">
-        <param name="name" value="mydomain.com"/>
-        <param name="login" value="myjid at myserver.com/talk"/>
-        <param name="password" value="mypass"/>
-        <param name="dialplan" value="XML"/>
-        <param name="message" value="Jingle all the way"/>
-        <param name="rtp-ip" value="10.0.0.1"/>
-        <param name="auto-login" value="true"/>
+	<param name="name" value="mydomain.com"/>
+	<param name="login" value="myjid at myserver.com/talk"/>
+	<param name="password" value="mypass"/>
+	<param name="dialplan" value="XML"/>
+	<param name="message" value="Jingle all the way"/>
+	<param name="rtp-ip" value="10.0.0.1"/>
+	<param name="auto-login" value="true"/>
 	<param name="auto-reply" value="Press *Call* to call FreeSWITCH and be sure to come to ClueCon! http://www.cluecon.com"/>
-        <!-- SASL "plain" or "md5" -->
-        <param name="sasl" value="plain"/>
-        <!-- if the server where the jabber is hosted is not the same as the one in the jid -->
-        <!--<param name="server" value="alternate.server.com"/>-->
-        <!-- Enable TLS or not -->
-        <param name="tls" value="true"/>
-        <!-- disable to trade async for more calls -->
-        <param name="use-rtp-timer" value="true"/>
-        <!-- or -->
-        <!-- <param name="rtp-ip" value="my_lan_ip"/> -->
-        <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/> -->
-        <!-- default extension (if one cannot be determined) -->
-        <param name="exten" value="888"/>
-        <!-- VAD choose one -->
-        <!-- <param name="vad" value="in"/> -->
-        <!-- <param name="vad" value="out"/> -->
-        <param name="vad" value="both"/>
+	<!-- SASL "plain" or "md5" -->
+	<param name="sasl" value="plain"/>
+	<!-- if the server where the jabber is hosted is not the same as the one in the jid -->
+	<!--<param name="server" value="alternate.server.com"/>-->
+	<!-- Enable TLS or not -->
+	<param name="tls" value="true"/>
+	<!-- disable to trade async for more calls -->
+	<param name="use-rtp-timer" value="true"/>
+	<!-- or -->
+	<!-- <param name="rtp-ip" value="my_lan_ip"/> -->
+	<!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/> -->
+	<!-- default extension (if one cannot be determined) -->
+	<param name="exten" value="888"/>
+	<!-- VAD choose one -->
+	<!-- <param name="vad" value="in"/> -->
+	<!-- <param name="vad" value="out"/> -->
+	<param name="vad" value="both"/>
       </x-profile>
 
       <!-- Component (Server to Server Login) -->
       <x-profile type="component">
 	<!-- All traffic for *@sub.mydomain.com will come to you -->
 	<param name="name" value="sub.mydomain.com"/>
-        <param name="password" value="secret"/>
+	<param name="password" value="secret"/>
 	<param name="dialplan" value="XML"/>
 	<param name="rtp-ip" value="208.64.200.42"/>
 	<param name="server" value="jabber.server.org:5347"/>
-        <!-- disable to trade async for more calls -->
-        <param name="use-rtp-timer" value="true"/>
+	<!-- disable to trade async for more calls -->
+	<param name="use-rtp-timer" value="true"/>
 	<!-- "_auto_" means the extension will be automaticly set to the called jid -->
-        <param name="exten" value="_auto_"/>
-        <!--<param name="vad" value="both"/>-->
+	<param name="exten" value="_auto_"/>
+	<!--<param name="vad" value="both"/>-->
       </x-profile>
 
     </configuration>
@@ -347,109 +347,131 @@
 
     <configuration name="xml_rpc.conf" description="XML RPC">
       <settings>
-        <!-- The port where you want to run the http service (default 8080) -->
-        <param name="http-port" value="8080"/>
-        <!-- if all 3 of the following params exist all http traffic will require auth -->
-        <param name="auth-realm" value="freeswitch"/>
-        <param name="auth-user" value="freeswitch"/>
-        <param name="auth-pass" value="works"/>
+	<!-- The port where you want to run the http service (default 8080) -->
+	<param name="http-port" value="8080"/>
+	<!-- if all 3 of the following params exist all http traffic will require auth -->
+	<param name="auth-realm" value="freeswitch"/>
+	<param name="auth-user" value="freeswitch"/>
+	<param name="auth-pass" value="works"/>
       </settings>
     </configuration>
 
     <configuration name="rss.conf" description="RSS Parser">
       <feeds>
-        <!-- Just download the files to wherever and refer to them here -->
-        <!-- <feed name="Slash Dot">/home/rss/rss.rss</feed> -->
-        <!-- <feed name="News Forge">/home/rss/newsforge.rss</feed> -->
+	<!-- Just download the files to wherever and refer to them here -->
+	<!-- <feed name="Slash Dot">/home/rss/rss.rss</feed> -->
+	<!-- <feed name="News Forge">/home/rss/newsforge.rss</feed> -->
       </feeds>
     </configuration>
 
     <!-- None of these paths are real if you want any of these options you need to really set them up -->
     <configuration name="conference.conf" description="Audio Conference">
-      <!-- Profiles are collections of settings you can reference by name. -->
+      <!-- Advertise certian presence on startup . -->
+      <advertise>
+	<room name="888 at sub.mydomain.com" status="FreeSWITCH"/>
+      </advertise>
 
+      <!-- Profiles are collections of settings you can reference by name. -->
       <profiles>
-        <profile name="default">
-          <!-- Sample Rate-->
-          <param name="rate" value="8000"/>
-          <!-- Number of milliseconds per frame -->
-          <param name="interval" value="20"/>
-          <!-- Energy level required for audio to be sent to the other users -->
-          <param name="energy-level" value="300"/>
-          <!-- TTS Engine to use -->
-          <!--<param name="tts-engine" value="cepstral"/>-->
-          <!-- TTS Voice to use -->
-          <!--<param name="tts-voice" value="david"/>-->
+	<profile name="default">
+	  <!-- Domain (for presence) -->
+	  <param name="domain" value="sub.mydomain.com"/>
+	  <!-- Sample Rate-->
+	  <param name="rate" value="8000"/>
+	  <!-- Number of milliseconds per frame -->
+	  <param name="interval" value="20"/>
+	  <!-- Energy level required for audio to be sent to the other users -->
+	  <param name="energy-level" value="300"/>
+	  <!-- TTS Engine to use -->
+	  <!--<param name="tts-engine" value="cepstral"/>-->
+	  <!-- TTS Voice to use -->
+	  <!--<param name="tts-voice" value="david"/>-->
 
-          <!-- If TTS is enabled all audio-file params not beginning with -->
-          <!-- '/' or with drive: (i.e. c:) will be considered text to say with TTS -->
+	  <!-- If TTS is enabled all audio-file params not beginning with -->
+	  <!-- '/' or with drive: (i.e. c:) will be considered text to say with TTS -->
 
-          <!-- File to play to acknowledge succees -->
-          <!--<param name="ack-sound" value="/soundfiles/beep.wav"/>-->
-          <!-- File to play to acknowledge failure -->
-          <!--<param name="nack-sound" value="/soundfiles/beeperr.wav"/>-->
-          <!-- File to play to acknowledge muted -->
-          <!--<param name="muted-sound" value="/soundfiles/muted.wav"/>-->
-          <!-- File to play to acknowledge unmuted -->
-          <!--<param name="unmuted-sound" value="/soundfiles/unmuted.wav"/>-->
-          <!-- File to play if you are alone in the conference -->
-          <!--<param name="alone-sound" value="/soundfiles/yactopitc.wav"/>-->
-          <!-- File to play when you join the conference -->
-          <!--<param name="enter-sound" value="/soundfiles/welcome.wav"/>-->
-          <!-- File to play when you leave the conference -->
-          <!--<param name="exit-sound" value="/soundfiles/exit.wav"/>-->
-          <!-- File to play when you ae ejected from the conference -->
-          <!--<param name="kicked-sound" value="/soundfiles/kicked.wav"/>-->
-          <!-- File to play when the conference is locked -->
-          <!--<param name="locked-sound" value="/soundfiles/locked.wav"/>-->
-          <!-- File to play to prompt for a pin -->
-          <!--<param name="pin-sound" value="/soundfiles/pin.wav"/>-->
-          <!-- File to play to when the pin is invalid -->
-          <!--<param name="bad-pin-sound" value="/soundfiles/invalid-pin.wav"/>-->
-          <!-- Conference pin -->
-          <!--<param name="pin" value="12345"/>-->
-          <!-- Default Caller ID Name for outbound calls -->
-          <param name="caller-id-name" value="FreeSWITCH"/>
-          <!-- Default Caller ID Number for outbound calls -->
-          <param name="caller-id-number" value="8777423583"/>
-        </profile>
+	  <!-- File to play to acknowledge succees -->
+	  <!--<param name="ack-sound" value="/soundfiles/beep.wav"/>-->
+	  <!-- File to play to acknowledge failure -->
+	  <!--<param name="nack-sound" value="/soundfiles/beeperr.wav"/>-->
+	  <!-- File to play to acknowledge muted -->
+	  <!--<param name="muted-sound" value="/soundfiles/muted.wav"/>-->
+	  <!-- File to play to acknowledge unmuted -->
+	  <!--<param name="unmuted-sound" value="/soundfiles/unmuted.wav"/>-->
+	  <!-- File to play if you are alone in the conference -->
+	  <!--<param name="alone-sound" value="/soundfiles/yactopitc.wav"/>-->
+	  <!-- File to play when you join the conference -->
+	  <!--<param name="enter-sound" value="/soundfiles/welcome.wav"/>-->
+	  <!-- File to play when you leave the conference -->
+	  <!--<param name="exit-sound" value="/soundfiles/exit.wav"/>-->
+	  <!-- File to play when you ae ejected from the conference -->
+	  <!--<param name="kicked-sound" value="/soundfiles/kicked.wav"/>-->
+	  <!-- File to play when the conference is locked -->
+	  <!--<param name="locked-sound" value="/soundfiles/locked.wav"/>-->
+	  <!-- File to play to prompt for a pin -->
+	  <!--<param name="pin-sound" value="/soundfiles/pin.wav"/>-->
+	  <!-- File to play to when the pin is invalid -->
+	  <!--<param name="bad-pin-sound" value="/soundfiles/invalid-pin.wav"/>-->
+	  <!-- Conference pin -->
+	  <!--<param name="pin" value="12345"/>-->
+	  <!-- Default Caller ID Name for outbound calls -->
+	  <param name="caller-id-name" value="FreeSWITCH"/>
+	  <!-- Default Caller ID Number for outbound calls -->
+	  <param name="caller-id-number" value="8777423583"/>
+	</profile>
       </profiles>
     </configuration>
 
     <configuration name="enum.conf" description="ENUM Module">
       <settings>
-				<param name="default-root" value="e164.org"/>
+	<param name="default-root" value="e164.org"/>
       </settings>
 
       <routes>
-				<route service="E2U+SIP" regex="sip:(.*)" replace="sofia/test/$1"/>
-				<route service="E2U+IAX2" regex="iax2:(.*)" replace="iax/$1"/>
-				<route service="E2U+XMPP" regex="XMPP:(.*)" replace="dingaling/jingle/$1"/>
+	<route service="E2U+SIP" regex="sip:(.*)" replace="sofia/test/$1"/>
+	<route service="E2U+IAX2" regex="iax2:(.*)" replace="iax/$1"/>
+	<route service="E2U+XMPP" regex="XMPP:(.*)" replace="dingaling/jingle/$1"/>
       </routes>
     </configuration>
 
     <configuration name="ivr.conf" description="IVR menus">
-			<menus>
-				<menu name="main" greet-long="/soundfiles/greet-long.wav" greet-short="/soundfiles/greet-short.wav" invalid-sound="/soundfiles/invalid.wav" exit-sound="/soundfiles/exit.wav" timeout ="15" max-failures="3">
-					<entry action="menu-exit" digits="*"/>
-					<entry action="menu-sub" digits="2" param="menu2"/>
-					<entry action="menu-exec-api" digits="3" param="api arg"/>
-					<entry action="menu-play-sound" digits="4" param="/soundfiles/4.wav"/>
-					<entry action="menu-back" digits="5"/>
-					<entry action="menu-call-transfer" digits="7" param="888"/>
-					<entry action="menu-sub" digits="8" param="menu8"/>>
-				</menu>
-				<menu name="menu8" greet-long="/soundfiles/greet-long.wav" greet-short="/soundfiles/greet-short.wav" invalid-sound="/soundfiles/invalid.wav" exit-sound="/soundfiles/exit.wav" timeout ="15" max-failures="3">
-					<entry action="menu-back" digits="#"/>
-					<entry action="menu-play-sound" digits="4" param="/soundfiles/4.wav"/>
-					<entry action="menu-top" digits="*"/>
-				</menu>
-				<menu name="menu2" greet-long="/soundfiles/greet-long.wav" greet-short="/soundfiles/greet-short.wav" invalid-sound="/soundfiles/invalid.wav" exit-sound="/soundfiles/exit.wav" timeout ="15" max-failures="3">
-					<entry action="menu-back" digits="#"/>
-					<entry action="menu-play-sound" digits="4" param="/soundfiles/4.wav"/>
-					<entry action="menu-top" digits="*"/>
-				</menu>
-			</menus>
+      <menus>
+	<menu name="main"
+	      greet-long="/soundfiles/greet-long.wav" 
+	      greet-short="/soundfiles/greet-short.wav"
+	      invalid-sound="/soundfiles/invalid.wav"
+	      exit-sound="/soundfiles/exit.wav" timeout ="15" max-failures="3">
+	  <entry action="menu-exit" digits="*"/>
+	  <entry action="menu-sub" digits="2" param="menu2"/>
+	  <entry action="menu-exec-api" digits="3" param="api arg"/>
+	  <entry action="menu-play-sound" digits="4" param="/soundfiles/4.wav"/>
+	  <entry action="menu-back" digits="5"/>
+	  <entry action="menu-call-transfer" digits="7" param="888"/>
+	  <entry action="menu-sub" digits="8" param="menu8"/>>
+	</menu>
+	<menu name="menu8"
+	      greet-long="/soundfiles/greet-long.wav"
+	      greet-short="/soundfiles/greet-short.wav"
+	      invalid-sound="/soundfiles/invalid.wav"
+	      exit-sound="/soundfiles/exit.wav"
+	      timeout ="15"
+	      max-failures="3">
+	  <entry action="menu-back" digits="#"/>
+	  <entry action="menu-play-sound" digits="4" param="/soundfiles/4.wav"/>
+	  <entry action="menu-top" digits="*"/>
+	</menu>
+	<menu name="menu2"
+	      greet-long="/soundfiles/greet-long.wav"
+	      greet-short="/soundfiles/greet-short.wav"
+	      invalid-sound="/soundfiles/invalid.wav"
+	      exit-sound="/soundfiles/exit.wav"
+	      timeout ="15"
+	      max-failures="3">
+	  <entry action="menu-back" digits="#"/>
+	  <entry action="menu-play-sound" digits="4" param="/soundfiles/4.wav"/>
+	  <entry action="menu-top" digits="*"/>
+	</menu>
+      </menus>
     </configuration> 
 
   </section>
@@ -462,47 +484,47 @@
     <!-- *NOTE* The special context name 'any' will match any context -->
     <context name="default">
       <extension name="tollfree">
-        <condition field="destination_number" expression="^(18(0{2}|8{2}|7{2}|6{2})\d{7})$">
+	<condition field="destination_number" expression="^(18(0{2}|8{2}|7{2}|6{2})\d{7})$">
 	  <action application="enum" data="$1"/>
 	  <action application="bridge" data="${enum_auto_route}"/>
-        </condition>
+	</condition>
       </extension>
 
       <!-- Call the FreeSWITCH conference via SIP -->
-        <!--<extension name="FreeSWITCH Conference SIP">-->
-          <!--<condition field="destination_number" expression="^888$">-->
-            <!--<action application="bridge" data="sofia/test/888 at conference.freeswitch.org"/>-->
-          <!--</condition>-->
-        <!--</extension> -->
+      <!--<extension name="FreeSWITCH Conference SIP">-->
+      <!--<condition field="destination_number" expression="^888$">-->
+      <!--<action application="bridge" data="sofia/test/888 at conference.freeswitch.org"/>-->
+      <!--</condition>-->
+      <!--</extension> -->
 
       <!-- Call the FreeSWITCH conference via IAX -->
-        <!--<extension name="FreeSWITCH Conference IAX">-->
-          <!--<condition field="destination_number" expression="^8888$">-->
-            <!--<action application="bridge" data="iax/guest at conference.freeswitch.org/888"/>-->
-          <!--</condition>-->
-        <!--</extension>-->
+      <!--<extension name="FreeSWITCH Conference IAX">-->
+      <!--<condition field="destination_number" expression="^8888$">-->
+      <!--<action application="bridge" data="iax/guest at conference.freeswitch.org/888"/>-->
+      <!--</condition>-->
+      <!--</extension>-->
 
       <extension name="testmusic">
-        <condition field="destination_number" expression="^1234$">
+	<condition field="destination_number" expression="^1234$">
 	  <!-- Request a certain tone/file to be played while you wait for the call to be answered-->
 	  <action application="set" data="ringback=${us-ring}"/>
 	  <!--<action application="set" data="ringback=/home/ring.wav"/>-->
-          <action application="bridge" data="sofia/test/1234 at conference.freeswitch.org"/>
-        </condition>
+	  <action application="bridge" data="sofia/test/1234 at conference.freeswitch.org"/>
+	</condition>
       </extension>
 
       <!-- Enter an existing conference -->
       <extension name="1000">
-        <condition field="destination_number" expression="^1000$">
-          <action application="conference" data="freeswitch"/>
-        </condition>
+	<condition field="destination_number" expression="^1000$">
+	  <action application="conference" data="freeswitch"/>
+	</condition>
       </extension>
 
       <!-- Start a dynamic conference and call someone at the same time -->
       <extension name="2000">
-        <condition field="destination_number" expression="^2000$">
-          <action application="conference" data="bridge:mydynaconf:sofia/test/1234 at conference.freeswitch.org"/>
-        </condition>
+	<condition field="destination_number" expression="^2000$">
+	  <action application="conference" data="bridge:mydynaconf:sofia/test/1234 at conference.freeswitch.org"/>
+	</condition>
       </extension>
 
       <!-- extensions starting with 4, all the numbers after 4 form a numeric filename -->
@@ -515,33 +537,33 @@
       <!-- the desired file.  This is for demo purposes both actions -->
       <!-- could have been under the same <extension> tag as well. -->
       <extension name="playsound1" continue="true">
-        <condition field="source" expression="mod_sofia"/>
-        <condition field="destination_number" expression="^4(\d+)">
-          <action application="playback" data="/var/sounds/beep.gsm"/>
-        </condition>
+	<condition field="source" expression="mod_sofia"/>
+	<condition field="destination_number" expression="^4(\d+)">
+	  <action application="playback" data="/var/sounds/beep.gsm"/>
+	</condition>
       </extension>
 
       <extension name="playsound2">
-        <condition field="source" expression="mod_sofia"/>
-        <condition field="destination_number" expression="^4(\d+)">
-          <action application="playback" data="/root/$1.raw"/>
-        </condition>
+	<condition field="source" expression="mod_sofia"/>
+	<condition field="destination_number" expression="^4(\d+)">
+	  <action application="playback" data="/root/$1.raw"/>
+	</condition>
       </extension>
 
       <!-- send everything with a certian RDNIS to Wanpipe ISDN -->
       <extension name="To PRI">
-        <condition field="rdnis" expression="8881231234"/>
-        <condition field="destination_number" expression="(.*)">
-          <action application="bridge" data="wanpipe/a/a/$1"/>
-        </condition>
+	<condition field="rdnis" expression="8881231234"/>
+	<condition field="destination_number" expression="(.*)">
+	  <action application="bridge" data="wanpipe/a/a/$1"/>
+	</condition>
       </extension>
 
       <!-- Call *MUST* originate from mod_iax and also be dialing ext 9999-->
       <extension name="9999">
-        <condition field="source" expression="mod_iax"/>
-        <condition field="destination_number" expression="9999">
-          <action application="playback" data="/var/sounds/beep.gsm"/>
-        </condition>
+	<condition field="source" expression="mod_iax"/>
+	<condition field="destination_number" expression="9999">
+	  <action application="playback" data="/var/sounds/beep.gsm"/>
+	</condition>
       </extension>
 
     </context>

Modified: freeswitch/branches/knhor/trunk/libs/libdingaling/.update
==============================================================================
--- freeswitch/branches/knhor/trunk/libs/libdingaling/.update	(original)
+++ freeswitch/branches/knhor/trunk/libs/libdingaling/.update	Wed Dec 13 13:04:49 2006
@@ -1 +1 @@
-Fri Dec  1 21:13:04 EST 2006
+Tue Dec 12 11:55:02 EST 2006

Modified: freeswitch/branches/knhor/trunk/libs/libdingaling/src/libdingaling.c
==============================================================================
--- freeswitch/branches/knhor/trunk/libs/libdingaling/src/libdingaling.c	(original)
+++ freeswitch/branches/knhor/trunk/libs/libdingaling/src/libdingaling.c	Wed Dec 13 13:04:49 2006
@@ -701,6 +701,8 @@
 	
 	if (type && !strcasecmp(type, "unavailable")) {
 		signal = LDL_SIGNAL_PRESENCE_OUT;
+    } else if (type && !strcasecmp(type, "probe")) {
+		signal = LDL_SIGNAL_PRESENCE_PROBE;
 	} else {
 		signal = LDL_SIGNAL_PRESENCE_IN;
 	}
@@ -725,20 +727,15 @@
 	}
 
 
-	
 	if (resource && strstr(resource, "talk") && (buffer = apr_hash_get(handle->probe_hash, id, APR_HASH_KEY_STRING))) {
 		apr_cpystrn(buffer->buf, from, buffer->len);
 		fflush(stderr);
 		buffer->hit = 1;
 	}
 	
-	if (!type || (type && strcasecmp(type, "probe"))) {
-
-		if (handle->session_callback) {
-			handle->session_callback(handle, NULL, signal, to, id, status ? status : "n/a", show ? show : "n/a");
-		}
-	}
-
+    if (handle->session_callback) {
+        handle->session_callback(handle, NULL, signal, to, id, status ? status : "n/a", show ? show : "n/a");
+    }
 
 	return IKS_FILTER_EAT;
 }

Modified: freeswitch/branches/knhor/trunk/libs/libdingaling/src/libdingaling.h
==============================================================================
--- freeswitch/branches/knhor/trunk/libs/libdingaling/src/libdingaling.h	(original)
+++ freeswitch/branches/knhor/trunk/libs/libdingaling/src/libdingaling.h	Wed Dec 13 13:04:49 2006
@@ -125,6 +125,7 @@
 	LDL_SIGNAL_MSG,
 	LDL_SIGNAL_PRESENCE_IN,
 	LDL_SIGNAL_PRESENCE_OUT,
+	LDL_SIGNAL_PRESENCE_PROBE,
 	LDL_SIGNAL_ROSTER,
 	LDL_SIGNAL_SUBSCRIBE,
 	LDL_SIGNAL_UNSUBSCRIBE,

Modified: freeswitch/branches/knhor/trunk/src/mod/applications/mod_conference/mod_conference.c
==============================================================================
--- freeswitch/branches/knhor/trunk/src/mod/applications/mod_conference/mod_conference.c	(original)
+++ freeswitch/branches/knhor/trunk/src/mod/applications/mod_conference/mod_conference.c	Wed Dec 13 13:04:49 2006
@@ -1357,7 +1357,7 @@
 				write_frame.data = data;
 				use_buffer = member->mux_buffer;
 
-				while ((write_frame.datalen = (uint32_t)switch_buffer_read(use_buffer, write_frame.data, bytes))) {
+				if ((write_frame.datalen = (uint32_t)switch_buffer_read(use_buffer, write_frame.data, bytes))) {
 					if (write_frame.datalen && switch_test_flag(member, MFLAG_CAN_HEAR)) {
 						write_frame.samples = write_frame.datalen / 2;
 
@@ -1370,10 +1370,11 @@
 				}
 
 				switch_mutex_unlock(member->audio_out_mutex);
+				continue;
 			}
-        }
+		}
 
-        switch_core_timer_next(&timer);
+		switch_core_timer_next(&timer);
 	} /* Rinse ... Repeat */
 
 	if (member->digit_stream != NULL) {

Modified: freeswitch/branches/knhor/trunk/src/mod/endpoints/mod_dingaling/mod_dingaling.c
==============================================================================
--- freeswitch/branches/knhor/trunk/src/mod/endpoints/mod_dingaling/mod_dingaling.c	(original)
+++ freeswitch/branches/knhor/trunk/src/mod/endpoints/mod_dingaling/mod_dingaling.c	Wed Dec 13 13:04:49 2006
@@ -2107,7 +2107,7 @@
 				}
 
 				if (is_special(to)) {
-					ldl_handle_send_presence(profile->handle, to, from, NULL, "unknown", "Click To Call");
+					ldl_handle_send_presence(profile->handle, to, from, NULL, NULL, "Click To Call");
 				}
 
 #if 0
@@ -2116,7 +2116,7 @@
 						switch_event_add_header(event, SWITCH_STACK_BOTTOM, "proto", MDL_CHAT_PROTO);
 						switch_event_add_header(event, SWITCH_STACK_BOTTOM, "login", "%s", profile->login);
 						switch_event_add_header(event, SWITCH_STACK_BOTTOM, "from", "%s",  to);
-						switch_event_add_header(event, SWITCH_STACK_BOTTOM, "rpid", "unknown");
+						//switch_event_add_header(event, SWITCH_STACK_BOTTOM, "rpid", "unknown");
 						switch_event_add_header(event, SWITCH_STACK_BOTTOM, "status", "Click To Call");
 						switch_event_fire(&event);
 					}
@@ -2130,6 +2130,11 @@
 					switch_event_fire(&event);
 				}
 				break;
+			case LDL_SIGNAL_PRESENCE_PROBE:
+				if (is_special(to)) {
+                    ldl_handle_send_presence(profile->handle, to, from, NULL, NULL, "Click To Call");
+                }
+                break;
 			case LDL_SIGNAL_PRESENCE_IN:
 				
 				if ((sql = switch_mprintf("update subscriptions set show='%q', status='%q' where sub_from='%q'", msg, subject, from))) {
@@ -2148,7 +2153,7 @@
 
 				
 				if (is_special(to)) {
-					ldl_handle_send_presence(profile->handle, to, from, NULL, "unknown", "Click To Call");
+					ldl_handle_send_presence(profile->handle, to, from, NULL, NULL, "Click To Call");
 				}
 #if 0
 				if (is_special(to)) {
@@ -2156,13 +2161,14 @@
 						switch_event_add_header(event, SWITCH_STACK_BOTTOM, "proto", MDL_CHAT_PROTO);
 						switch_event_add_header(event, SWITCH_STACK_BOTTOM, "login", "%s", profile->login);
 						switch_event_add_header(event, SWITCH_STACK_BOTTOM, "from", "%s",  to);
-						switch_event_add_header(event, SWITCH_STACK_BOTTOM, "rpid", "unknown");
+						//switch_event_add_header(event, SWITCH_STACK_BOTTOM, "rpid", "unknown");
 						switch_event_add_header(event, SWITCH_STACK_BOTTOM, "status", "Click To Call");
 						switch_event_fire(&event);
 					}
 				}
 				break;
 #endif
+
 			case LDL_SIGNAL_PRESENCE_OUT:
 				
 				if ((sql = switch_mprintf("update subscriptions set show='%q', status='%q' where sub_from='%q'", msg, subject, from))) {

Modified: freeswitch/branches/knhor/trunk/src/mod/endpoints/mod_portaudio/mod_portaudio.c
==============================================================================
--- freeswitch/branches/knhor/trunk/src/mod/endpoints/mod_portaudio/mod_portaudio.c	(original)
+++ freeswitch/branches/knhor/trunk/src/mod/endpoints/mod_portaudio/mod_portaudio.c	Wed Dec 13 13:04:49 2006
@@ -149,8 +149,12 @@
 	if (switch_test_flag(tech_pvt, TFLAG_OUTBOUND)) {
 
 		switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "%s CHANNEL INIT %d %d\n", switch_channel_get_name(channel),
-			switch_channel_get_state(channel), switch_test_flag(tech_pvt, TFLAG_ANSWER));
+                          switch_channel_get_state(channel), switch_test_flag(tech_pvt, TFLAG_ANSWER));
+        
 
+
+        engage_device(tech_pvt);
+
 		while (switch_channel_get_state(channel) == CS_INIT && !switch_test_flag(tech_pvt, TFLAG_ANSWER)) {
 			if (switch_time_now() - last >= waitsec) {
 				char buf[512];
@@ -159,8 +163,8 @@
 				snprintf(buf, sizeof(buf), "BRRRRING! BRRRRING! call %s\n", tech_pvt->call_id);
 
 				if (switch_event_create_subclass(&event, SWITCH_EVENT_CUSTOM, MY_EVENT_RINGING) == SWITCH_STATUS_SUCCESS) {
-					switch_event_add_header(event, SWITCH_STACK_BOTTOM, "event_info", buf);
-					switch_event_add_header(event, SWITCH_STACK_BOTTOM, "call_id", tech_pvt->call_id);
+					switch_event_add_header(event, SWITCH_STACK_BOTTOM, "event_info", "%s", buf);
+					switch_event_add_header(event, SWITCH_STACK_BOTTOM, "call_id", "%s", tech_pvt->call_id);
 					switch_channel_event_set_data(channel, event);
 					switch_event_fire(&event);
 				}
@@ -171,10 +175,6 @@
 			switch_yield(50000);
 		}
 		
-		if (switch_channel_ready(channel)) {
-			/* Turn on the device */
-			engage_device(tech_pvt);
-		}
 	}
 
 



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