[asterisk-scf-commits] asterisk-scf/integration/test_channel.git branch "rename" created.
Commits to the Asterisk SCF project code repositories
asterisk-scf-commits at lists.digium.com
Thu Dec 29 11:00:26 CST 2011
branch "rename" has been created
at 2df267b26a355a6ebc743fda8d437c685e89d49e (commit)
- Log -----------------------------------------------------------------
commit 2df267b26a355a6ebc743fda8d437c685e89d49e
Author: Mark Michelson <mmichelson at digium.com>
Date: Tue Dec 27 16:34:35 2011 -0600
Adjust for logging name changes.
diff --git a/src/CMakeLists.txt b/src/CMakeLists.txt
index d0b4800..f8ae737 100644
--- a/src/CMakeLists.txt
+++ b/src/CMakeLists.txt
@@ -18,7 +18,7 @@ astscf_component_add_ice_libraries(test_channel IceStorm)
astscf_component_add_boost_libraries(test_channel thread date_time)
astscf_component_add_slice_collection_libraries(test_channel ASTSCF TEST_CHANNEL)
astscf_component_build_icebox(test_channel)
-target_link_libraries(test_channel logging-client)
+target_link_libraries(test_channel LoggingClient)
astscf_component_install(test_channel)
astscf_component_init(console_driver)
commit 0117d4b9d993671d0eb84b38df579f7a55c6aa61
Author: Mark Michelson <mmichelson at digium.com>
Date: Mon Oct 17 09:17:32 2011 -0500
Adjust for party id changes.
diff --git a/src/TestEndpoint.cpp b/src/TestEndpoint.cpp
index 0b0315c..5c9c7bc 100644
--- a/src/TestEndpoint.cpp
+++ b/src/TestEndpoint.cpp
@@ -231,7 +231,8 @@ public:
NamePtr name = new Name("bar");
NumberPtr number = new Number("100");
- IdPtr testId = new Id(name, number);
+ PrivacyPtr privacy = new Privacy(false);
+ IdPtr testId = new Id(name, number, privacy);
IdSeq idSeq;
idSeq.push_back(testId);
mCaller = new Caller(idSeq);
@@ -240,20 +241,23 @@ public:
NamePtr dialedName = new Name("foo");
NumberPtr dialedNumber = new Number("104");
- IdPtr dialedId = new Id(dialedName, dialedNumber);
+ PrivacyPtr dialedPrivacy = new Privacy(false);
+ IdPtr dialedId = new Id(dialedName, dialedNumber, dialedPrivacy);
IdSeq dialedSeq;
dialedSeq.push_back(dialedId);
mDialed = new Dialed(dialedNumber);
NamePtr redirName = new Name("scud");
NumberPtr redirNumber = new Number("666");
- IdPtr connectedId = new Id(redirName, redirNumber);
+ PrivacyPtr redirPrivacy = new Privacy(false);
+ IdPtr connectedId = new Id(redirName, redirNumber, redirPrivacy);
IdSeq idSeq2;
idSeq2.push_back(connectedId);
mConnectedLine = new ConnectedLine(idSeq2);
RedirectionSeq redirects;
- RedirectionPtr redirect = new Redirection(dialedId, connectedId);
+ RedirectionReasonPtr reason = new RedirectionReason(Unknown);
+ RedirectionPtr redirect = new Redirection(dialedId, connectedId, reason);
redirects.push_back(redirect);
mRedirections = new Redirections(redirects);
commit ecdc524619fb486d95d9aded350f935344f434e0
Author: Ken Hunt <ken.hunt at digium.com>
Date: Sun Oct 2 18:11:11 2011 -0500
Added Telephony Event sink and source support.
diff --git a/src/TestEndpoint.cpp b/src/TestEndpoint.cpp
index 18fd07c..0b0315c 100644
--- a/src/TestEndpoint.cpp
+++ b/src/TestEndpoint.cpp
@@ -20,6 +20,7 @@
#include <Ice/Ice.h>
#include <IceUtil/UUID.h>
+#include <AsteriskSCF/Helpers/ProxyHelper.h>
#include <AsteriskSCF/SessionCommunications/SessionCommunicationsIf.h>
#include <boost/thread/locks.hpp>
@@ -131,15 +132,95 @@ protected:
typedef IceUtil::Handle<SessionListenerMgr> SessionListenerMgrPtr;
+class TelephonyEventSinkI : public AsteriskSCF::SessionCommunications::V1::TelephonyEventSink
+{
+public:
+ TelephonyEventSinkI() {}
+
+ void write_async(
+ const AsteriskSCF::SessionCommunications::V1::AMD_TelephonyEventSink_writePtr& cb,
+ const AsteriskSCF::SessionCommunications::V1::TelephonyEventPtr& telephonyEvent,
+ const Ice::Current&)
+ {
+ mEvents.push_back(telephonyEvent);
+ cb->ice_response();
+ }
+
+ void setSource_async(
+ const AsteriskSCF::SessionCommunications::V1::AMD_TelephonyEventSink_setSourcePtr& cb,
+ const AsteriskSCF::SessionCommunications::V1::TelephonyEventSourcePrx& source,
+ const Ice::Current&)
+ {
+ mSource = source;
+ cb->ice_response();
+ }
+
+
+ void getSource_async(
+ const AsteriskSCF::SessionCommunications::V1::AMD_TelephonyEventSink_getSourcePtr& cb,
+ const Ice::Current&)
+ {
+ cb->ice_response(mSource);
+ }
+
+private:
+
+ AsteriskSCF::SessionCommunications::V1::TelephonyEventSourcePrx mSource;
+ std::vector<AsteriskSCF::SessionCommunications::V1::TelephonyEventPtr> mEvents;
+};
+typedef IceUtil::Handle<TelephonyEventSinkI> TelephonyEventSinkIPtr;
+
+class TelephonyEventSourceI : public AsteriskSCF::SessionCommunications::V1::TelephonyEventSource
+{
+public:
+
+ TelephonyEventSourceI() {}
+
+ // TelephonyEventSource API implementation...
+
+ void addSinks_async(
+ const AsteriskSCF::SessionCommunications::V1::AMD_TelephonyEventSource_addSinksPtr& cb,
+ const AsteriskSCF::SessionCommunications::V1::TelephonyEventSinkSeq& sinks,
+ const Ice::Current&)
+ {
+ mSinks.insert(mSinks.end(), sinks.begin(), sinks.end());
+ cb->ice_response();
+ }
+
+ void removeSinks_async(
+ const AsteriskSCF::SessionCommunications::V1::AMD_TelephonyEventSource_removeSinksPtr& cb,
+ const AsteriskSCF::SessionCommunications::V1::TelephonyEventSinkSeq& sinks,
+ const Ice::Current&)
+ {
+ for (TelephonyEventSinkSeq::const_iterator i = sinks.begin(); i != sinks.end(); ++i)
+ {
+ mSinks.erase(std::remove_if(mSinks.begin(), mSinks.end(), AsteriskSCF::IdentityComparePredicate<TelephonyEventSinkPrx>(*i)), mSinks.end());
+ }
+ cb->ice_response();
+ }
+
+ void getSinks_async(
+ const AsteriskSCF::SessionCommunications::V1::AMD_TelephonyEventSource_getSinksPtr& cb,
+ const Ice::Current&)
+ {
+ cb->ice_response(mSinks);
+ }
+
+private:
+ AsteriskSCF::SessionCommunications::V1::TelephonyEventSinkSeq mSinks;
+};
+typedef IceUtil::Handle<TelephonyEventSourceI> TelephonyEventSourceIPtr;
+
//
// Needs some intelligence:
// - scheduled ringing
// - etc.
-class SessionI : public AsteriskSCF::SessionCommunications::V1::Session
+class SessionI : public AsteriskSCF::SessionCommunications::V1::TelephonySession
{
public:
SessionI(const InternalManagerPtr& m, const AsteriskSCF::SessionCommunications::V1::SessionEndpointPrx& prx, const std::string& id,
const Ice::ObjectAdapterPtr& adapter) :
+ mAdapter(adapter),
mEndpointManager(m),
mEndpointPrx(prx),
mId(id),
@@ -175,6 +256,15 @@ public:
RedirectionPtr redirect = new Redirection(dialedId, connectedId);
redirects.push_back(redirect);
mRedirections = new Redirections(redirects);
+
+ mTelephonyEventSinkPtr = new TelephonyEventSinkI();
+ TelephonyEventSinkPrx sinkPrx = TelephonyEventSinkPrx::uncheckedCast(mAdapter->addWithUUID(mTelephonyEventSinkPtr));
+ mTelephonyEventSinks.push_back(sinkPrx);
+
+ mTelephonyEventSourcePtr = new TelephonyEventSourceI();
+ TelephonyEventSourcePrx sourcePrx = TelephonyEventSourcePrx::uncheckedCast(mAdapter->addWithUUID(mTelephonyEventSourcePtr));
+ mTelephonyEventSources.push_back(sourcePrx);
+
}
void addListener_async(
@@ -419,8 +509,36 @@ public:
cb->ice_response(mRedirections);
}
+
+ //// TelephonySession overrides
+
+ void getSources_async(
+ const AsteriskSCF::SessionCommunications::V1::AMD_TelephonySession_getSourcesPtr& cb,
+ const Ice::Current&)
+ {
+ cb->ice_response(mTelephonyEventSources);
+ }
+
+ void getSinks_async(
+ const AsteriskSCF::SessionCommunications::V1::AMD_TelephonySession_getSinksPtr& cb,
+ const Ice::Current&)
+ {
+ cb->ice_response(mTelephonyEventSinks);
+ }
+
+ TelephonyEventSourceIPtr getSource()
+ {
+ return mTelephonyEventSourcePtr;
+ }
+
+ TelephonyEventSinkIPtr getSink()
+ {
+ return mTelephonyEventSinkPtr;
+ }
+
private:
boost::shared_mutex mMutex;
+ Ice::ObjectAdapterPtr mAdapter;
InternalManagerPtr mEndpointManager;
AsteriskSCF::SessionCommunications::V1::SessionEndpointPrx mEndpointPrx;
AsteriskSCF::SessionCommunications::V1::SessionInfoPtr mInfo;
@@ -429,11 +547,17 @@ private:
AsteriskSCF::Media::V1::SessionPrx mMediaSession;
SessionListenerMgrPtr mListeners;
AsteriskSCF::SessionCommunications::V1::BridgePrx mCurrentBridge;
+
SessionOwnerIdPtr mSessionOwnerId;
CallerPtr mCaller;
DialedPtr mDialed;
ConnectedLinePtr mConnectedLine;
RedirectionsPtr mRedirections;
+
+ TelephonyEventSourceIPtr mTelephonyEventSourcePtr;
+ TelephonyEventSinkIPtr mTelephonyEventSinkPtr;
+ TelephonyEventSourceSeq mTelephonyEventSources;
+ TelephonyEventSinkSeq mTelephonyEventSinks;
};
typedef IceUtil::Handle<SessionI> SessionIPtr;
commit cd0b4f5a9dc0b7e753f28160eaacb6457fc7477b
Author: Ken Hunt <ken.hunt at digium.com>
Date: Fri Sep 30 17:29:09 2011 -0500
Adapt to changes in Party Id API.
diff --git a/src/TestEndpoint.cpp b/src/TestEndpoint.cpp
index 1dec56f..18fd07c 100644
--- a/src/TestEndpoint.cpp
+++ b/src/TestEndpoint.cpp
@@ -171,7 +171,10 @@ public:
idSeq2.push_back(connectedId);
mConnectedLine = new ConnectedLine(idSeq2);
- mRedirecting = new Redirecting(dialedId, connectedId, 1);
+ RedirectionSeq redirects;
+ RedirectionPtr redirect = new Redirection(dialedId, connectedId);
+ redirects.push_back(redirect);
+ mRedirections = new Redirections(redirects);
}
void addListener_async(
@@ -411,9 +414,9 @@ public:
cb->ice_response(mConnectedLine);
}
- void getRedirecting_async(const AMD_Session_getRedirectingPtr& cb, const Ice::Current& )
+ void getRedirections_async(const AMD_Session_getRedirectionsPtr& cb, const Ice::Current& )
{
- cb->ice_response(mRedirecting);
+ cb->ice_response(mRedirections);
}
private:
@@ -430,7 +433,7 @@ private:
CallerPtr mCaller;
DialedPtr mDialed;
ConnectedLinePtr mConnectedLine;
- RedirectingPtr mRedirecting;
+ RedirectionsPtr mRedirections;
};
typedef IceUtil::Handle<SessionI> SessionIPtr;
-----------------------------------------------------------------------
--
asterisk-scf/integration/test_channel.git
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