This reverts commit bdf11be97bd732e8891ae19342c937da6e659afa.
Fix a missing import from manual merge.
Change-Id: If373626f07250cbfe07e5c04cf02ad9ee5a0ab2a
Merge commit 'edec27a1e9ffc022e68f0d6200ba90499da4b9e9'
* commit 'edec27a1e9ffc022e68f0d6200ba90499da4b9e9':
Do not explicity disconnect Data during power down for 1x.
Merge commit 'd531c9ebe18f0b554d29d3c3b8e4a00f84dae97a' into gingerbread-plus-aosp
* commit 'd531c9ebe18f0b554d29d3c3b8e4a00f84dae97a':
Do not explicity disconnect Data during power down for 1x.
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
Merge commit '1738252a596c71851cabf5835acb3584ad6b3191'
* commit '1738252a596c71851cabf5835acb3584ad6b3191':
Don't enter DISCONNECTING state when the call/connection is not alive
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 '4a04a3129bd30a996dd302b982aeca8f228f57e8'
* commit '4a04a3129bd30a996dd302b982aeca8f228f57e8':
Fix the unhold issue especially if one is behind NAT.