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authorkpfleming <kpfleming@f38db490-d61c-443f-a65b-d21fe96a405b>2008-07-28 16:49:18 +0000
committerkpfleming <kpfleming@f38db490-d61c-443f-a65b-d21fe96a405b>2008-07-28 16:49:18 +0000
commit02b6d501ac4ec8afa68343fd7eba07c871def467 (patch)
treec260b454ee36057daa09da1d6ddef150eb246521 /doc/osp.txt
parent8c653dc5c125614f66b4d353e2187e3e0c4ea215 (diff)
Merged revisions 134086 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r134086 | kpfleming | 2008-07-28 11:42:00 -0500 (Mon, 28 Jul 2008) | 3 lines remove remaining Zaptel references in various places ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@134087 f38db490-d61c-443f-a65b-d21fe96a405b
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@@ -304,7 +304,7 @@ defaultprotocol=SIP
3.1.3 extensions.conf
OSP functions are implemented as dial plan functions in the extensions.conf file. To add OSP support to your Asterisk server, simply copy and paste the text box below to your extensions.conf file. These functions will enable your Asterisk server to support all OSP call scenarios. Configuration of your Asterisk server for OSP is now complete.
[globals]
-DIALOUT=Zap/1
+DIALOUT=DAHDI/1
[SrcGW] ; OSP Source Gateway
exten => _XXXX.,1,NoOp(OSPSrcGW)