aboutsummaryrefslogtreecommitdiffstatshomepage
path: root/bin/OpenSim.ini.example
diff options
context:
space:
mode:
authorSean Dague2008-04-15 14:24:15 +0000
committerSean Dague2008-04-15 14:24:15 +0000
commit6f8ff326307ae1522e2f3163596b0bf1cdd2157f (patch)
tree4517fcd0560e9cf05ad170c0032fc9a60e7a01e4 /bin/OpenSim.ini.example
parentFrom: dirk husemann <hud@zurich.ibm.com> (diff)
downloadopensim-SC_OLD-6f8ff326307ae1522e2f3163596b0bf1cdd2157f.zip
opensim-SC_OLD-6f8ff326307ae1522e2f3163596b0bf1cdd2157f.tar.gz
opensim-SC_OLD-6f8ff326307ae1522e2f3163596b0bf1cdd2157f.tar.bz2
opensim-SC_OLD-6f8ff326307ae1522e2f3163596b0bf1cdd2157f.tar.xz
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...
Diffstat (limited to '')
-rw-r--r--bin/OpenSim.ini.example16
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
173; Global SIP Server for conference calls 173; Global SIP Server for conference calls
174sip_domain = testserver.com 174sip_domain = testserver.com
175 175
176[AsteriskVoice]
177; PLEASE NOTE that we don't have voice support in OpenSim quite yet - these configuration options are stubs
178enabled = false
179; SIP account server domain
180sip_domain = testserver.com
181; SIP conf server domain
182conf_domain = testserver.com
183; URL of the asterisk opensim frontend
184asterisk_frontend = http://testserver.com:49153/
185; password for the asterisk frontend XmlRpc calls
186asterisk_password = bah-humbug
187; timeout for XmlRpc calls to asterisk front end (in ms)
188asterisk_timeout = 3000
189; salt for asterisk nonces
190asterisk_salt = paluempalum
191
176; Uncomment the following to control the progression of daytime 192; Uncomment the following to control the progression of daytime
177; in the Sim. The defaults are what is shown below 193; in the Sim. The defaults are what is shown below
178;[Sun] 194;[Sun]