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, 5 insertions, 0 deletions

diff --git a/share/python/asterisk/create-user.sql b/share/python/asterisk/create-user.sql
new file mode 100644
index 0000000..68b8147
--- /dev/null
+++ b/share/python/asterisk/create-user.sql
@@ -0,0 +1,5 @@
+USE %(database)s;
+REPLACE INTO ast_sipfriends (port,context,disallow,allow,type,secret,host,name) VALUES ('5060','avatare','all','ulaw','friend','%(password)s','dynamic','%(username)s');
+REPLACE INTO `extensions_table` (context,exten,priority,app,appdata) VALUES ('avatare', '%(username)s', 1, 'Answer', '');
+REPLACE INTO `extensions_table` (context,exten,priority,app,appdata) VALUES ('avatare', '%(username)s', 2, 'Wait', '1');
+REPLACE INTO `extensions_table` (context,exten,priority,app,appdata) VALUES ('avatare', '%(username)s', 3, 'Dial', 'SIP/%(username)s,60');
\ No newline at end of file