OpenSimulator - User contributions [en]

Freeswitch Module

2009-05-01T22:35:10Z

Joelmorrissette: /* configure conference.conf.xml */

The FreeSwitch module enables voice in opensim with no changes required to the SL or Hippo clients (must be over 1.22 for SL and 0.5 for Hippo)

== FreeSwitch Install ==

Follow the instructions [http://wiki.freeswitch.org/wiki/Installation_Guide here] on how to compile from source. We need to enable two specific modules.
'''
please ensure you compile from the freeswitch trunk for now until we can post a minimum version number (there are known issues with older versions)'''

When you get to the part in the instructions where it says "'''Edit modules.conf so that it will build the modules you desire.'''" edit the modules.conf file and uncomment out the entries for xml_curl and the siren14 codec

codecs/mod_siren
and
xml_int/mod_xml_curl

* You can also use this [http://madhawa.com/?p=10 guide] for setting it up in a quick way -- [[User:Fly-man-|Fly-man-]]

==FreeSwitch Config==

Install and compile Freeswitch, making sure you enable the xml_curl module and also the siren14 codec.

=== enable mod_xml_curl ===

Next, do not forget to activate mod_xml_curl in /usr/local/freeswitch/conf/autoload_configs/modules.conf.xml. mod_xml_curl is disabled by default on fresh install.

uncomment the lines...
<load module="mod_xml_curl"/>

and

<load module="mod_siren"/>

are uncommented

=== configure mod_xml_curl ===

the xml_curl module configuration should point to an opensim region that has the freeswitch voice module enabled (voice also needs to be enabled in the estate setting for all regions)

example xml_curl.conf.xml found in /usr/local/freeswitch/conf/autoload_configs

<configuration name="xml_curl.conf" description="cURL XML Gateway">
<bindings>
<binding name="example">
<param name="gateway-url" value="http://youropensimregion:9000/api/freeswitch-config" bindings="directory"/>
<param name="disable-100-continue" value="true"/>
</binding>
<binding name="local">
<param name="gateway-url" value="http://youropensimregion:9000/api/freeswitch-config" bindings="dialplan"/>
<param name="disable-100-continue" value="true"/>
</binding>
</bindings>
</configuration>

The /usr/local/freeswitch/conf/vars.xml requires modification to enable the siren14 codec

search within vars.xml for the global_codec_prefs and change the line to read

<X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221@32000h,G722,PCMU,PCMA,GSM"/>

G7221@32000h is the siren14 codec

=== configure conference.conf.xml ===
By default, FreeSwitch plays hold music when there is only one avatar in the conference and beeps for everyone when avatars arrive and leave. To disable, edit /usr/local/freeswitch/conf/autoload_configs/conference.conf.xml. Locate the "default" profile and comment out the following lines as shown below:

[...]


[...]








[...]

==OpenSim Config==

Add the following section to OpenSim.ini. You will also need to enable voice in the regions estate settings. Make sure the freeswitch server is started BEFORE bringing the region up.

[FreeSwitchVoice]
enabled = true
;FreeSwitch server is going to contact us and ask us all
;sorts of things.
freeswitch_server_user = freeswitch
freeswitch_server_pass = password
freeswitch_api_prefix = /api
;The IP address of your opensim voice region
freeswitch_service_server = youropensimexternalIP
;the port your region is running on
freeswitch_service_port = 9000
;your freewitch IP address
freeswitch_realm = 192.168.0.2
freeswitch_sip_proxy = 192.168.0.2:5060
freeswitch_attempt_stun = false
freeswitch_stun_server = 192.168.0.2
freeswitch_echo_server = 192.168.0.2
freeswitch_echo_port = 50505
freeswitch_well_known_ip = 192.168.0.2
freeswitch_default_timeout = 5000
freeswitch_subscribe_retry = 120

Freeswitch Module

2009-05-01T18:06:23Z

Joelmorrissette: /* configure conference.conf.xml */

The FreeSwitch module enables voice in opensim with no changes required to the SL or Hippo clients (must be over 1.22 for SL and 0.5 for Hippo)

== FreeSwitch Install ==

Follow the instructions [http://wiki.freeswitch.org/wiki/Installation_Guide here] on how to compile from source. We need to enable two specific modules.
'''
please ensure you compile from the freeswitch trunk for now until we can post a minimum version number (there are known issues with older versions)'''

When you get to the part in the instructions where it says "'''Edit modules.conf so that it will build the modules you desire.'''" edit the modules.conf file and uncomment out the entries for xml_curl and the siren14 codec

codecs/mod_siren
and
xml_int/mod_xml_curl

* You can also use this [http://madhawa.com/?p=10 guide] for setting it up in a quick way -- [[User:Fly-man-|Fly-man-]]

==FreeSwitch Config==

Install and compile Freeswitch, making sure you enable the xml_curl module and also the siren14 codec.

=== enable mod_xml_curl ===

Next, do not forget to activate mod_xml_curl in /usr/local/freeswitch/conf/autoload_configs/modules.conf.xml. mod_xml_curl is disabled by default on fresh install.

uncomment the lines...
<load module="mod_xml_curl"/>

and

<load module="mod_siren"/>

are uncommented

=== configure mod_xml_curl ===

the xml_curl module configuration should point to an opensim region that has the freeswitch voice module enabled (voice also needs to be enabled in the estate setting for all regions)

example xml_curl.conf.xml found in /usr/local/freeswitch/conf/autoload_configs

<configuration name="xml_curl.conf" description="cURL XML Gateway">
<bindings>
<binding name="example">
<param name="gateway-url" value="http://youropensimregion:9000/api/freeswitch-config" bindings="directory"/>
<param name="disable-100-continue" value="true"/>
</binding>
<binding name="local">
<param name="gateway-url" value="http://youropensimregion:9000/api/freeswitch-config" bindings="dialplan"/>
<param name="disable-100-continue" value="true"/>
</binding>
</bindings>
</configuration>

The /usr/local/freeswitch/conf/vars.xml requires modification to enable the siren14 codec

search within vars.xml for the global_codec_prefs and change the line to read

<X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221@32000h,G722,PCMU,PCMA,GSM"/>

G7221@32000h is the siren14 codec

=== configure conference.conf.xml ===
By default, FreeSwitch plays hold music when there is only one avatar in the conference and beeps for everyone when avatars arrive and leave. To disable, edit /usr/local/freeswitch/conf/autoload_configs/conference.conf.xml. Locate the "default" profile and comment out the following lines as shown below:

[...]


[...]








[...]

By default, FreeSwitch records every conference to a wav file. Comment out the following line as shown below to disable this:



==OpenSim Config==

Add the following section to OpenSim.ini. You will also need to enable voice in the regions estate settings. Make sure the freeswitch server is started BEFORE bringing the region up.

[FreeSwitchVoice]
enabled = true
;FreeSwitch server is going to contact us and ask us all
;sorts of things.
freeswitch_server_user = freeswitch
freeswitch_server_pass = password
freeswitch_api_prefix = /api
;The IP address of your opensim voice region
freeswitch_service_server = youropensimexternalIP
;the port your region is running on
freeswitch_service_port = 9000
;your freewitch IP address
freeswitch_realm = 192.168.0.2
freeswitch_sip_proxy = 192.168.0.2:5060
freeswitch_attempt_stun = false
freeswitch_stun_server = 192.168.0.2
freeswitch_echo_server = 192.168.0.2
freeswitch_echo_port = 50505
freeswitch_well_known_ip = 192.168.0.2
freeswitch_default_timeout = 5000
freeswitch_subscribe_retry = 120

Freeswitch Module

2009-05-01T17:57:54Z

Joelmorrissette: /* configure conference.conf.xml */

Freeswitch Module

2009-05-01T17:54:01Z

Joelmorrissette: /* configure conference.conf.xml */

The FreeSwitch module enables voice in opensim with no changes required to the SL or Hippo clients (must be over 1.22 for SL and 0.5 for Hippo)

== FreeSwitch Install ==

Follow the instructions [http://wiki.freeswitch.org/wiki/Installation_Guide here] on how to compile from source. We need to enable two specific modules.
'''
please ensure you compile from the freeswitch trunk for now until we can post a minimum version number (there are known issues with older versions)'''

When you get to the part in the instructions where it says "'''Edit modules.conf so that it will build the modules you desire.'''" edit the modules.conf file and uncomment out the entries for xml_curl and the siren14 codec

codecs/mod_siren
and
xml_int/mod_xml_curl

* You can also use this [http://madhawa.com/?p=10 guide] for setting it up in a quick way -- [[User:Fly-man-|Fly-man-]]

==FreeSwitch Config==

Install and compile Freeswitch, making sure you enable the xml_curl module and also the siren14 codec.

=== enable mod_xml_curl ===

Next, do not forget to activate mod_xml_curl in /usr/local/freeswitch/conf/autoload_configs/modules.conf.xml. mod_xml_curl is disabled by default on fresh install.

uncomment the lines...
<load module="mod_xml_curl"/>

and

<load module="mod_siren"/>

are uncommented

=== configure mod_xml_curl ===

the xml_curl module configuration should point to an opensim region that has the freeswitch voice module enabled (voice also needs to be enabled in the estate setting for all regions)

example xml_curl.conf.xml found in /usr/local/freeswitch/conf/autoload_configs

<configuration name="xml_curl.conf" description="cURL XML Gateway">
<bindings>
<binding name="example">
<param name="gateway-url" value="http://youropensimregion:9000/api/freeswitch-config" bindings="directory"/>
<param name="disable-100-continue" value="true"/>
</binding>
<binding name="local">
<param name="gateway-url" value="http://youropensimregion:9000/api/freeswitch-config" bindings="dialplan"/>
<param name="disable-100-continue" value="true"/>
</binding>
</bindings>
</configuration>

The /usr/local/freeswitch/conf/vars.xml requires modification to enable the siren14 codec

search within vars.xml for the global_codec_prefs and change the line to read

<X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221@32000h,G722,PCMU,PCMA,GSM"/>

G7221@32000h is the siren14 codec

=== configure conference.conf.xml ===

Comment out the following lines as shown below within the file /usr/local/freeswitch/conf/autoload_configs/conference.conf.xml, if you want to disable music-on-hold when there is only one avatar in a freeswitch conference and to disable the beeps when new avatars arrive or leave:

[...]


[...]








[...]

==OpenSim Config==

Add the following section to OpenSim.ini. You will also need to enable voice in the regions estate settings. Make sure the freeswitch server is started BEFORE bringing the region up.

[FreeSwitchVoice]
enabled = true
;FreeSwitch server is going to contact us and ask us all
;sorts of things.
freeswitch_server_user = freeswitch
freeswitch_server_pass = password
freeswitch_api_prefix = /api
;The IP address of your opensim voice region
freeswitch_service_server = youropensimexternalIP
;the port your region is running on
freeswitch_service_port = 9000
;your freewitch IP address
freeswitch_realm = 192.168.0.2
freeswitch_sip_proxy = 192.168.0.2:5060
freeswitch_attempt_stun = false
freeswitch_stun_server = 192.168.0.2
freeswitch_echo_server = 192.168.0.2
freeswitch_echo_port = 50505
freeswitch_well_known_ip = 192.168.0.2
freeswitch_default_timeout = 5000
freeswitch_subscribe_retry = 120