Merge commit '0a69f597604254bc37721b135ab612eaacdd0cbd' into gingerbread-plus-aosp
* commit '0a69f597604254bc37721b135ab612eaacdd0cbd':
Rub in a little 'ol log-b-gone.
* Fix some typos in Javadoc and log messages.
* Remove redundant initializer in BluetoothAdapter.readOutOfBandData()
* Use canonical "UTF-8" charset name instead of "UTF8" in
BluetoothDevice.convertPinToBytes()
Change-Id: I58cd5dc48a7ad0053d204c5f590b4b3d438d8672
Merge commit '421c34c162098efe870574844a7ee49812bbb929' into gingerbread-plus-aosp
* commit '421c34c162098efe870574844a7ee49812bbb929':
SipPhone: revise hangup() in SipCall and SipConnection.
Exceptions may throw during canTake() as the peer may cancel the call and
result in a race with this method call.
Change-Id: I61903d601d8f9b2dcb4c4fbe1586e2c1a1069109
http://b/issue?id=3033868
Make them DISCONNECTED immediately. Don't enter DISCONNECTING state and wait
until SipSession ends the session. SipSession will get timed out eventually
but PhoneApp/user don't need to know this detail and wait.
This should fix the bug:
http://b/issue?id=3027719
Change-Id: Ida5a1bd09d08b9d591721384b4978127619aab51
CallerInfoAsyncQuery can now handle SIP addresses in addition to regular
phone numbers: if the number passed in to startQuery() is actually a "URI
number", we now treat it as a SIP address and look it up directly in the
Data table.
If it's a regular phone number, the behavior is unchanged: we use the
PhoneLookup table as before.
This piece of the fix covers only the contact lookup for incoming calls;
we still need some more cleanup of the CallerInfo class in order to get
the call log working.
Bug: 3004127
Change-Id: I0fcb80f9de5b8ecf99d31ee92e0889ddb07216fd
and fix how SipErrorCode.SERVER_ERROR is determinted from server response, not
from local exceptions.
http://b/issue?id=3041332
Change-Id: Idce67e29858d5c7573b98b7fa1fac074913d71d6
Merge commit '245475925eff61ee76bde58de69253a889e39d0a' into gingerbread-plus-aosp
* commit '245475925eff61ee76bde58de69253a889e39d0a':
Fix the startAudio order for 3-way calls.
Merge commit '3234652242f54e3366e7c74e5a0cf0a7da5871b4' into gingerbread-plus-aosp
* commit '3234652242f54e3366e7c74e5a0cf0a7da5871b4':
Don't enter DISCONNECTING state when the call/connection is not alive
+ check REQUEST_TERMINATED response on INVITE not CANCEL,
+ check if a TransactionTerminatedEvent matches the ongoing transaction,
+ add log to track SipConnection disconnect events.
Change-Id: I28325be62ac44e4a7507d3c4b5b78b066c0ea2ad
and add new CROSS_DOMAIN_AUTHENTICATION error code and OUT_OF_NETWORK
DisconnectCause.
http://b/issue?id=3020185
Change-Id: Icc0a341599d5a72b7cb2d43675fbddc516544978
Merge commit 'bd2294204e3edaede3fe81eb9b11c05c4fafe627' into gingerbread-plus-aosp
* commit 'bd2294204e3edaede3fe81eb9b11c05c4fafe627':
Fix the unhold issue especially if one is behind NAT.
Merge commit '194bbcce9ba15634500f542b9ea017b2cf154b45' into gingerbread-plus-aosp
* commit '194bbcce9ba15634500f542b9ea017b2cf154b45':
SIP: longer timeout for making call, shorter for cancelling
Merge commit '84a357bb6a8005e1c5e924e96a8ecf310e77c47c' into gingerbread-plus-aosp
* commit '84a357bb6a8005e1c5e924e96a8ecf310e77c47c':
Refactoring SIP classes to get ready for API review.
+ replace SipAudioCall and its Listener interfaces with real implementations,
+ remove SipAudioCallImpl.java, most of it is has become part of SipAudioCall,
+ add SipSession and its Listener classes to wrap ISipSession and ISipSessionListener,
+ move SipSessionState to SipSession.State,
+ make SipManager keep context and remove the context argument from many methods of its,
+ rename SipManager.getInstance() to newInstance(),
+ rename constant names for action strings and extra keys to follow conventions,
+ set thread names for debugging purpose.
Change-Id: Ie1790dc0e8f49c06c7fc80d33fec0f673a9c3044
+fix the unknown call flash for answering an incoming call and
updating the screen if the background call got dropped.
+change the getFirstActiveBgCall to return the call if the state
is not IDLE. This will help to fix unknown flash if the background
call got dropped.
Change-Id: I9803ccebd919acbd5296e7dfde7dc5f29cc9f180
Merge commit 'ee2ef3220fd27a6332acb2f65951a7fe91e9dfa6' into gingerbread-plus-aosp
* commit 'ee2ef3220fd27a6332acb2f65951a7fe91e9dfa6':
Use PhoneBase in the phone list.
also fix delivering bad news before closing a SipAudioCallImpl object so that
apps can get the current audio-call object state before it's closed:
http://b/issue?id=3009262
Change-Id: I94c19dae8b4f252de869e614ec462b19b4ff2077
Merge commit '97963794af1e18674dd111e3ad344d90b16c922c' into gingerbread-plus-aosp
* commit '97963794af1e18674dd111e3ad344d90b16c922c':
SIP: convert enum to static final int.
For bug 3001613.
Only use PhoneBase (not PhoneProxy) in CallManager.
Both PhoneBase and PhoneProxy implement Phone interface,
such as dial(). The real implementation, for
example in GSM, is in GSMPhone extending from PhoneBase.
So that foregroundCall.getPhone() returns GSMPhone obj. On the other hand,
PhoneFactory.getDefaultPhone() returns PhoneProxy obj, which has a class
member of GSMPhone.
Therefore for phone returned by PhoneFacotry, which is used by PhoneApp,
phone.getForegroundCall().getPhone() != phone
Change-Id: I8a304098dd86762aaee56fb3c8b76c883e8c9a4f
Merge commit '1d1583573d2099756bbbeef48d97c280edc393e0' into gingerbread-plus-aosp
* commit '1d1583573d2099756bbbeef48d97c280edc393e0':
SipPhone: do not append SIP domain to PSTN number
