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| author | Kevin P. Fleming <kpfleming@digium.com> | 2008-07-28 16:42:00 +0000 |
|---|---|---|
| committer | Kevin P. Fleming <kpfleming@digium.com> | 2008-07-28 16:42:00 +0000 |
| commit | 6291cd19bff749a1492b4b5601ed814b886f11c4 (patch) | |
| tree | a7c0f5a4a58e8458f72230691577432e5fe36cb1 /doc/osp.txt | |
| parent | 06d951f58588d6713a71d2aa689a07e2af70254e (diff) | |
remove remaining Zaptel references in various places
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@134086 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Diffstat (limited to 'doc/osp.txt')
| -rw-r--r-- | doc/osp.txt | 2 |
1 files changed, 1 insertions, 1 deletions
diff --git a/doc/osp.txt b/doc/osp.txt index 6bad3a25c..a0cd42968 100644 --- a/doc/osp.txt +++ b/doc/osp.txt @@ -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) |