Adding a simple API enabling applications to control SMS-CB reception.
Implementing parsing, assembly and dispatching of SMS-CB messages over GSM.
Change-Id: Iee841605a45a3af60c7602af175056afb03a38da
Fix bug # 3136179.
Keep audio mode as IN_CALL during hangup DISCONNECTING state
to prevent the NORMAL and IN_CALL glitch in auiod setMode.
Change-Id: I5513a3d5c65bd13ac054c9718c4dbd7d6db9eaf3
instead of silently returning null and causing NPE in applications as returning
null is not documented in the javadoc.
Add connection to the connection list in SipCall after dial() succeeds so that
we don't need to clean up if it fails. The original code will cause the failed
connection to continue to live in the SipCall and in next dial() attempt, a new
connection is created and the in-call screen sees two connections in the call
and thus shows conference call UI.
Bug: 3157234, 3157387
Change-Id: Iabc3235f781c4f1e09384a67ad56b09ad2c12e5e
In order to reduce object creation the RILRequest objects are
stored in an array when it is unused (max 4). This avoids GC
of the object. The object in turn has references to other
objects which sometimes hold large memory chunks. This fix
releases these references since they are not used anyway.
This will make it possible to GC the Message (mResult) which
in some cases holds references to a Bitmap which sometimes
leads to OutOfMemoryException. The reference is cleared
anyway in RILRequest.obtain(...)
Change-Id: I3b895bc39b5e2f3ab7cc8297c3583ea78e0ebc77
When a SIP call is put on hold and no other call is active, the audio mode should not be
switched to incall.
Change-Id: I1307330f10cbfb9c4223bcb9dc4faa79778750af
Merge commit '6fe795ecd35c4d49822d349424fc71b660577dfc' into gingerbread-plus-aosp
* commit '6fe795ecd35c4d49822d349424fc71b660577dfc':
Do another contact lookup if the first one fails and...
Merge commit 'baced375ba5f374445c44a2115700d69693794a0' into gingerbread-plus-aosp
* commit 'baced375ba5f374445c44a2115700d69693794a0':
Silently reject a ringing call when another call is dialing/ringing.
Merge commit '1180f2a099a134c40f923c7e4162a5e7d7ca0184' into gingerbread-plus-aosp
* commit '1180f2a099a134c40f923c7e4162a5e7d7ca0184':
Remove ringtone API from SipAudioCall.
+ Avoid concurrent modification when forming >3-way conf call.
+ Revise SipConnection.separate() to put the newly separated call to foreground.
Bug: 3114987
Change-Id: If6204e7e3cc05f4a516c33657a368b53a0ad014d
it's a SIP call and the peer's username is all numeric. The all-numeric username
could be a PSTN number.
Bug: 3105116 (case #2)
Change-Id: I1de9cfac3aab1c4c89935176264d07693adb5e7d
(watch out auto-merge conflict for SipAudioCall).
Bug: 3113033, related CL: https://android-git/g/#change,75185
Change-Id: Ib48d3b990e229e0b341e47e10e76934e1a50d10f
Merge commit 'b595e094e3901ff8a224eaf7d4869e7d2e5649dd' into gingerbread-plus-aosp
* commit 'b595e094e3901ff8a224eaf7d4869e7d2e5649dd':
Return display name in SipConnection.getCnapName().
Remember, the system and main logs are
- Shared resources
- Primarily for recording problems
- To be used only for large grained events during normal operation
Bug: 3104855
Change-Id: I136fbd101917dcbc8ebc3f96f276426b48bde7b7
Merge commit 'f5201ab71ff4d104265ab126e86afc6b81da8011' into gingerbread-plus-aosp
* commit 'f5201ab71ff4d104265ab126e86afc6b81da8011':
Keep original phone number in SipConnection.
In case it's a PSTN number carried by an Internet call, the phone app can still
get the original phone number from Connection.getAddress() instead of getting a
SIP URI.
http://b/issue?id=3085996
Change-Id: Ie6c66100a4b5b2ce3f73baa1b446761cd51d7727
Merge commit 'c8511af04a442551a204b1f47fabb317bcf54be0' into gingerbread-plus-aosp
* commit 'c8511af04a442551a204b1f47fabb317bcf54be0':
Add mock ril control commands and tests
Merge commit '0e430ccc2c8a4bb9d96002676d7742652bd28477' into gingerbread-plus-aosp
* commit '0e430ccc2c8a4bb9d96002676d7742652bd28477':
Telephony: Fix radio state printing
Currently the SipPhone class manually creates a CallerInfo object, and
populates it with very basic info from the SIP address, when making an
outgoing call.
But this is no longer needed, now that we do caller-id lookup properly for
SIP addresses (based on real data from the contacts database -- see
bug 3004127 and change https://android-git.corp.google.com/g/70555).
And in fact the presence of this initial CallerInfo object actually
*disabled* contacts lookup for outgoing calls (bug 3072731).
This change removes all that CallerInfo-related stuff from SipPhone.
(Thus SipPhone is now consistent with the other phone objects, like
GSMPhone and CDMAPhone, in that it doesn't muck with CallerInfo data at
all, but instead lets the phone app do it.)
Also, update isUriNumber() to handle "%40" in case the passed-in string is
URI-escaped. (Nobody depends on that now, but it may be needed in the
future, and it's certainly safe to say that "%40" will never be found in a
legal PSTN number.)
TESTED:
- Outgoing SIP call:
- In-call UI shows correct contact info
- After the call, Call Log shows correct contact info
- Incoming SIP call:
- In-call UI shows correct contact info
- After the call, Call Log shows correct contact info
- PSTN calls:
- correct contact info everywhere
Bug: 3072731
Change-Id: I51434e4e5ad66d2e8ff51fc220001fb74485f0f5
Add mock ril controller commands and test cases:
- testStartIncomingCallAndHangup: test start incoming cal and hangup remote
- testSetCallTransitionFlag: test call transition flag and call state transition
Change-Id: I25ff8ef7931159ef7101b5e8638b9b7438db4f66
NullPointerException at
com.android.internal.telephony.gsm.GSMPhone.handleMessage(GSMPhone.java)
Failing to retrieved the IMSI number from SIM card could lead to
an exception. A null pointer check will prevent this.
Change-Id: I26760543484504c8d35215bfb1e8f1ae664aeade
