From: Dr Scofield <hud@zurich.ibm.com>

   ansgar and i have been working on an asterisk voice module that will allow
   us to couple opensim with an asterisk VoIP gateway.

   the patch below consists of
     * AsteriskVoiceModule region module: alternative to the plain-vanilla
       VoiceModule, will make XmlRpc calls out to an asterisk-opensim
       frontend
     * asterisk-opensim.py frontend, living in share/python/asterisk, takes
       XmlRpc calls from the AsteriskVoiceModule
          * account_update: to update/create a new SIP account (on
            ProvisionVoiceAccountRequest)
          * region_update: to update/create a new "region" conference call
            (on ParcelVoiceInfo)
     * a asterisk-opensim test client, living in share/python/asterisk, to
       exercise astersik-opensim.py
   this still does not give us voice in OpenSim, but it's another step on
   this path...

1 files changed, 16 insertions, 0 deletions

diff --git a/bin/OpenSim.ini.example b/bin/OpenSim.ini.example
index 6a04947..4e99740 100644
--- a/bin/OpenSim.ini.example
+++ b/bin/OpenSim.ini.example
@@ -173,6 +173,22 @@ account_management_server = https://www.bhr.vivox.com/api2
 ; Global SIP Server for conference calls
 sip_domain = testserver.com
 
+[AsteriskVoice]
+; PLEASE NOTE that we don't have voice support in OpenSim quite yet - these configuration options are stubs
+enabled = false
+; SIP account server domain 
+sip_domain = testserver.com
+; SIP conf server domain 
+conf_domain = testserver.com
+; URL of the asterisk opensim frontend 
+asterisk_frontend = http://testserver.com:49153/
+; password for the asterisk frontend XmlRpc calls
+asterisk_password = bah-humbug
+; timeout for XmlRpc calls to asterisk front end (in ms)
+asterisk_timeout = 3000
+; salt for asterisk nonces
+asterisk_salt = paluempalum
+
 ; Uncomment the following to control the progression of daytime
 ; in the Sim.  The defaults are what is shown below
 ;[Sun]