[asterisk-scf-commits] asterisk-scf/integration/slice.git branch "bridgetelephony" created.
Commits to the Asterisk SCF project code repositories
asterisk-scf-commits at lists.digium.com
Wed Sep 7 17:15:14 CDT 2011
branch "bridgetelephony" has been created
at 08dd418b601ddb9836e6bbb912a4d0210fd2622c (commit)
- Log -----------------------------------------------------------------
commit 08dd418b601ddb9836e6bbb912a4d0210fd2622c
Author: Ken Hunt <ken.hunt at digium.com>
Date: Wed Sep 7 17:12:17 2011 -0500
Modified the TelephonyEventSource interface. Changed addSink() to addSinks() to cut down on RPC calls. Added a removeSinks() operation.
diff --git a/slice/AsteriskSCF/SessionCommunications/TelephonyEventsIf.ice b/slice/AsteriskSCF/SessionCommunications/TelephonyEventsIf.ice
index 11acb78..33031a9 100644
--- a/slice/AsteriskSCF/SessionCommunications/TelephonyEventsIf.ice
+++ b/slice/AsteriskSCF/SessionCommunications/TelephonyEventsIf.ice
@@ -71,9 +71,15 @@ module V1
interface TelephonyEventSource
{
/**
- * Add a new sink to send telephony events to
+ * Add sinks to send telephony events to.
*/
- ["amd"] void addSink(TelephonyEventSink* sink);
+ ["amd"] void addSinks(TelephonyEventSinkSeq sinks);
+
+ /**
+ * Remove sinks.
+ */
+ ["amd"] void removeSinks(TelephonyEventSinkSeq sinks);
+
/**
* Get a list of the sinks to which this source sends events
*/
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asterisk-scf/integration/slice.git
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