bye bye iax

git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@16487 d0543943-73ff-0310-b7d9-9358b9ac24b2

diff --git a/conf/autoload_configs/enum.conf.xml b/conf/autoload_configs/enum.conf.xml
index eb7a37de794e0f9266cc0f9b572600b5402782ad..0304d0dacd4fd9eda9e6d9afcfe4c14f5f59ed63 100644 (file)
--- a/conf/autoload_configs/enum.conf.xml
+++ b/conf/autoload_configs/enum.conf.xml
@@ -8,7 +8,6 @@
 
   <routes>
     <route service="E2U+SIP" regex="sip:(.*)" replace="sofia/${use_profile}/$1"/>
-    <!--<route service="E2U+IAX2" regex="iax2:(.*)" replace="iax/$1"/>-->
     <!--<route service="E2U+XMPP" regex="XMPP:(.*)" replace="dingaling/$${xmpp_server_profile}/$1"/>-->
   </routes>
 </configuration>

diff --git a/conf/autoload_configs/iax.conf.xml b/conf/autoload_configs/iax.conf.xml
deleted file mode 100644 (file)
index 3346369..0000000
--- a/conf/autoload_configs/iax.conf.xml
+++ /dev/null
@@ -1,14 +0,0 @@
-<configuration name="iax.conf" description="IAX Configuration">
-  <settings>
-    <param name="debug" value="0"/>
-    <param name="ip" value="$${local_ip_v4}"/>
-    <param name="port" value="4569"/>
-    <param name="context" value="public"/>
-    <param name="dialplan" value="XML"/>
-    <param name="codec-prefs" value="PCMU@20i,PCMA,speex,L16"/>
-    <param name="codec-master" value="us"/>
-    <param name="codec-rates" value="8"/>
-    <!-- seconds till we give up when there is no media -->
-    <param name="media-timeout" value="300"/>
-  </settings>
-</configuration>

diff --git a/conf/autoload_configs/modules.conf.xml b/conf/autoload_configs/modules.conf.xml
index 0781ade128d75ccd60e68f5ff7647e8a9d08e89f..803cd309ee10baa53e80c1930c078fbb48fa8caf 100644 (file)
--- a/conf/autoload_configs/modules.conf.xml
+++ b/conf/autoload_configs/modules.conf.xml
@@ -28,7 +28,6 @@
 
     <!-- Endpoints -->
     <!-- <load module="mod_dingaling"/> -->
-    <!-- <load module="mod_iax"/> -->
     <!-- <load module="mod_portaudio"/> -->
     <!-- <load module="mod_alsa"/> -->
     <load module="mod_sofia"/>

diff --git a/debian/freeswitch.conffiles b/debian/freeswitch.conffiles
index e8fc1d87609b78f72ef63f32421c19ed54e29b96..1869b5bd7a009d2aeb3498d3ee91856890736075 100644 (file)
--- a/debian/freeswitch.conffiles
+++ b/debian/freeswitch.conffiles
@@ -69,7 +69,6 @@
 /opt/freeswitch/conf/autoload_configs/event_socket.conf.xml
 /opt/freeswitch/conf/autoload_configs/modules.conf.xml
 /opt/freeswitch/conf/autoload_configs/dingaling.conf.xml
-/opt/freeswitch/conf/autoload_configs/iax.conf.xml
 /opt/freeswitch/conf/autoload_configs/post_load_modules.conf.xml
 /opt/freeswitch/conf/autoload_configs/pocketsphinx.conf.xml
 /opt/freeswitch/conf/autoload_configs/xml_cdr.conf.xml

diff --git a/debian/freeswitch.install b/debian/freeswitch.install
index 741d66d7b8b573064e38cd0c632637a75dec53b5..ee08d7a18f2bb05bfa05e34d333c538b8362e397 100644 (file)
--- a/debian/freeswitch.install
+++ b/debian/freeswitch.install
@@ -41,7 +41,6 @@ opt/freeswitch/mod/mod_voipcodecs.so*
 opt/freeswitch/mod/mod_speex.so*
 opt/freeswitch/mod/mod_dialplan*.so*
 opt/freeswitch/mod/mod_dingaling.so*
-opt/freeswitch/mod/mod_iax.so*
 opt/freeswitch/mod/mod_portaudio.so*
 opt/freeswitch/mod/mod_sofia.so*
 opt/freeswitch/mod/mod_openzap.so
@@ -141,7 +140,6 @@ opt/freeswitch/conf/autoload_configs/conference.conf.xml
 opt/freeswitch/conf/autoload_configs/event_socket.conf.xml
 opt/freeswitch/conf/autoload_configs/modules.conf.xml
 opt/freeswitch/conf/autoload_configs/dingaling.conf.xml
-opt/freeswitch/conf/autoload_configs/iax.conf.xml
 opt/freeswitch/conf/autoload_configs/post_load_modules.conf.xml
 opt/freeswitch/conf/autoload_configs/pocketsphinx.conf.xml
 opt/freeswitch/conf/autoload_configs/xml_cdr.conf.xml

diff --git a/debian/rules b/debian/rules
index 3dc5e51e00c69cd14ac81042c61543a886a5253e..abf3d784f87b1c428f477d770e52a88efadf6bce 100755 (executable)
--- a/debian/rules
+++ b/debian/rules
@@ -20,7 +20,7 @@ export CODECS_MODULES=codecs/mod_ilbc codecs/mod_h26x codecs/mod_speex codecs/mo
 export DIALPLANS_MODULES=dialplans/mod_dialplan_asterisk dialplans/mod_dialplan_directory dialplans/mod_dialplan_xml
 export DIRECTORIES_MODULES=
 export DOTNET_MODULES=
-export ENDPOINTS_MODULES=endpoints/mod_dingaling endpoints/mod_iax endpoints/mod_portaudio endpoints/mod_sofia \
+export ENDPOINTS_MODULES=endpoints/mod_dingaling endpoints/mod_portaudio endpoints/mod_sofia \
                         endpoints/mod_loopback ../../libs/openzap/mod_openzap endpoints/mod_skypiax
 export EVENT_HANDLERS_MODULES=event_handlers/mod_event_multicast event_handlers/mod_event_socket event_handlers/mod_cdr_csv
 export FORMATS_MODULES=formats/mod_local_stream formats/mod_native_file formats/mod_sndfile formats/mod_tone_stream formats/mod_shout

diff --git a/freeswitch.spec b/freeswitch.spec
index 4857cf55d0686aac8724a9539d2c879cda824b97..6c42da530a86aee21efe1b4970b044e91049ec9e 100644 (file)
--- a/freeswitch.spec
+++ b/freeswitch.spec
@@ -78,7 +78,7 @@ and chat driven products scaling from a soft-phone up to a soft-switch.  It can
 simple switching engine, a media gateway or a media server to host IVR applications using 
 simple scripts or XML to control the callflow. 
 
-We support various communication technologies such as SIP, H.323, IAX2 and GoogleTalk making 
+We support various communication technologies such as SIP, H.323 and GoogleTalk making 
 it easy to interface with other open source PBX systems such as sipX, OpenPBX, Bayonne, YATE or Asterisk.
 
 We also support both wide and narrow band codecs making it an ideal solution to bridge legacy 
@@ -232,7 +232,7 @@ APPLICATIONS_MODULES="applications/mod_commands applications/mod_conference appl
 CODECS_MODULES="codecs/mod_ilbc codecs/mod_h26x codecs/mod_voipcodecs codecs/mod_speex codecs/mod_celt codecs/mod_siren codecs/mod_bv"
 DIALPLANS_MODULES="dialplans/mod_dialplan_asterisk dialplans/mod_dialplan_directory dialplans/mod_dialplan_xml"
 DIRECTORIES_MODULES=""
-ENDPOINTS_MODULES="endpoints/mod_dingaling endpoints/mod_iax endpoints/mod_portaudio endpoints/mod_sofia ../../libs/openzap/mod_openzap endpoints/mod_loopback"
+ENDPOINTS_MODULES="endpoints/mod_dingaling endpoints/mod_portaudio endpoints/mod_sofia ../../libs/openzap/mod_openzap endpoints/mod_loopback"
 ASR_TTS_MODULES="asr_tts/mod_pocketsphinx asr_tts/mod_flite asr_tts/mod_unimrcp"
 EVENT_HANDLERS_MODULES="event_handlers/mod_event_multicast event_handlers/mod_event_socket event_handlers/mod_cdr_csv"
 FORMATS_MODULES="formats/mod_local_stream formats/mod_native_file formats/mod_sndfile formats/mod_tone_stream formats/mod_shout"
@@ -394,7 +394,6 @@ fi
 %config(noreplace) %attr(0640, freeswitch, daemon) %{prefix}/conf/autoload_configs/fifo.conf.xml
 %config(noreplace) %attr(0640, freeswitch, daemon) %{prefix}/conf/autoload_configs/shout.conf.xml
 %config(noreplace) %attr(0640, freeswitch, daemon) %{prefix}/conf/autoload_configs/timezones.conf.xml
-%config(noreplace) %attr(0640, freeswitch, daemon) %{prefix}/conf/autoload_configs/iax.conf.xml
 %config(noreplace) %attr(0640, freeswitch, daemon) %{prefix}/conf/autoload_configs/ivr.conf.xml
 %config(noreplace) %attr(0640, freeswitch, daemon) %{prefix}/conf/autoload_configs/java.conf.xml
 %config(noreplace) %attr(0640, freeswitch, daemon) %{prefix}/conf/autoload_configs/limit.conf.xml
@@ -473,7 +472,6 @@ fi
 %{prefix}/mod/mod_dialplan_xml.so* 
 %{prefix}/mod/mod_dialplan_asterisk.so* 
 %{prefix}/mod/mod_dingaling.so* 
-%{prefix}/mod/mod_iax.so* 
 %{prefix}/mod/mod_portaudio.so* 
 %{prefix}/mod/mod_sofia.so* 
 %{prefix}/mod/mod_cdr_csv.so*