If we're asked to connect to a DUN APN and we have a carrier specified
DUN APN setting, verify what we're connected to is the same as what the
carrier specified before accepting it.
bug:4048013
Change-Id: I91edc4a1342cb40c1f6959e149303b7d76710f96
1. Database changes:
- Add a protocol and a roaming_protocol column to the
carriers table in the telephony provider database.
- Set the protocol and roaming_protocol fields when
creating APN objects from the database.
2. ApnSetting class changes:
- Add protocol and roamingProtocol fields to the
ApnSetting class that encapsulates APN settings within
the framework.
- Add the fields to ApnSetting.toString and support a new
syntax containing the fields in ApnSetting.fromString.
- Add a unit test for ApnSetting.
3. Telephony changes:
- Specify the APN protocol when setting up a data call,
using protocol when not roaming and roaming_protocol
when roaming.
This change depends on #86896 in the telephony provider,
which adds the new column to the database schema on
upgrades.
Bug: 3333633
Change-Id: If3d9ed4c851d0192849df0d64581db03b066e052
Backport the protocol changes to setupDataCall and its
callers from master. As in master, hardcode IPv4
connectivity for now. When we add the protocol field to
ApnSettings, it will be fetched from there.
Bug: 3333633
Change-Id: I51880bc0ec192cbf964ac7bbd6a4b7d2eed41d27
'*86' which is set now as a default at CDMAPhone.java,
is default voicemail number for Verizon.
For Sprint, we use user's own number for voicemail.
So we add codes in CDMAPhone.java to use
'config_telephony_use_own_number_for_voicemail',
and use config.xml to set this value as false.
Then we overlay Sprint's own config.xml file to
override 'config_telephony_use_own_number_for_voicemail' as a true.
Change-Id: I110914bdfa9a79aaba89d3b80edbcf044e9aabee
Normally ADN record is stored in two tags, EFADN_TAG and EFEXT1_TAG. But if the EFEXT1_TAG is not set in EF_PBR entry,
it shouldn't be decoded otherwise it causes NullPointerException in readAdnFileAndWait().
bug: 3376954
Change-Id: I744cf5cb8662c75be1e7c6219fa92c67bdda82f2
If users dial 92-96, dial them normally and not treat
as USSD
Change-Id: If3b6cb37b7ec0ff99d76cb10cba53368094a0b5d
Signed-off-by: sj2202.park@samsung.com
Default Android MNC value has a 2 digit but it should be supported a 3 digit
MNC in India. (should be supported both 2 and 3 digits MNC)
Change-Id: I69373d196b29bccd06653841f24cbfe3886834fb
Signed-off-by: Sang-Jun Park <sj2202.park@samsung.com>
TTY mode should not be restricted to CDMA phones as some GSM carriers
support it.
TTY support is enabled by overlaying the tty_enabled boolean property
in packages/apps/Phones/res/values/config.xml
Also corrected wrong comments on TTY methods.
Change-Id: I48dbc2be51c3dcdaedc1838b85134edc7012be3c
1. According to TS 23.040, TP-Status values is changed properly.
2. When processing Status Report, it should be checked whether tpStatus is PENDING or FAILED.
Change-Id: I91c315cfb363f3e4b936c6b6b1a01083687a580f
Adding them hidden so that if OEM's are rolling their own at least they can
use the same values. Will mark them unhidden in a future sdk release.
bug:3395729
Change-Id: I90eabe036a96e1aa7c8cac49ca51efd9b1776a0c
Cherripick from master CL 79833, 79417, 78864, 80332, 87500
Add new audio mode and recording source for audio communications
other than telelphony.
The audio mode MODE_IN_CALL signals the system the device a phone
call is currently underway. There was no way for audio video
chat or VoIP applications to signal a call is underway, but not
using the telephony resources. This change introduces a new mode
to address this. Changes in other parts of the system (java
and native) are required to take this new mode into account.
The generic AudioPolicyManager is updated to not use its phone
state variable directly, but to use two new convenience methods,
isInCall() and isStateInCall(int) instead.
Add a recording source used to designate a recording stream for
voice communications such as VoIP.
Update the platform-independent audio policy manager to pass the
nature of the audio recording source to the audio policy client
interface through the AudioPolicyClientInterface::setParameters()
method.
SIP calls should set the audio mode to MODE_IN_COMMUNICATION,
Audio mode MODE_IN_CALL is reserved for telephony.
SIP: Enable built-in echo canceler if available.
1. Always initialize AudioRecord with VOICE_COMMUNICATION.
2. If echo canceler is available, disable our echo suppressor.
Note that this CL is intentionally not correcting the
getAudioSourceMax() return value in MediaRecorder.java as the
new source is hidden here.
Change-Id: Ie68cd03c50553101aa2ad838fe9459b2cf151bc8
Wakelock will get released while
1) no request pending to be sent out, in which mRequestMessagesPending increases
before calling EVENT_SEND and decreases while handling EVENT_SEND.
and
2) no waiting requests sent to RIL but no replied, in which mRequestMessagesWaiting
increases while sending request and decreases while handling response.
Both will be cleared while WAKE_LOCK_TIMEOUT occurs to recovery from out of sync situation.
bug: 3369427, 3370827
Change-Id: Ib2fc54db3b155bd3fb1296ad83720b7836708caf
Currently, PhoneUtils.getMute() returns the mute state from the foreground phone.
When a SIP call is muted and then put on hold, the call is moved to background
and the SipPhone becomes background phone. At this point, PhoneUtils.getMute()
incorrectly returns false from the idle foreground phone (i.e., GSMPhone).
CallManager provides getMute() but it's not used anywhere. This CL fixes the
method and I'll have another CL to have PhoneUtils.getMute() take advantage of
it.
Bug: 3323789
Change-Id: I6c37500ae93f4e95db3bcd55e24e1ecb58a57c0a
The wakelock will be kept held if there is outstanding requests
in request list. When WAKE_LOCK_TIMEOUT occurs, all requests
in mRequestList already waited at least DEFAULT_WAKE_LOCK_TIMEOUT
but no response. Those lost requests return GENERIC_FAILURE and
request list is cleared.
bug:3292426
Change-Id: I369c6ba4d6836d65ef616140e48c7304faf888f0
Rewrote interceptKeyBeforeQueueing to make the handling more systematic.
Behavior should be identical except:
- We never pass keys to applications when the screen is off and the keyguard
is not showing (the proximity sensor turned off the screen).
Previously we passed all non-wake keys through in this case which
caused a bug on Crespo where the screen would come back on if a soft key
was held at the time of power off because the resulting key up event
would sneak in just before the keyguard was shown. It would then be
passed through to the dispatcher which would poke user activity and
wake up the screen.
- We propagate the key flags when broadcasting media keys which
ensures that recipients can tell when the key is canceled.
- We ignore endcall or power if canceled (shouldn't happen anyways).
Changed the input dispatcher to not poke user activity for canceled
events since they are synthetic and should not wake the device.
Changed the lock screen so that it does not poke the wake lock when the
grab handle is released. This fixes a bug where the screen would come
back on immediately if the power went off while the user was holding
one of the grab handles because the sliding tab would receive an up
event after screen turned off and release the grab handles.
Bug: 3144874
Change-Id: Iebb91e10592b4ef2de4b1dd3a2e1e4254aacb697
For bug #3164802.
CallManager allow a new phone call only if ALL of the following are true:
- Phone is not powered off
- There's no incoming or waiting call
- There's available call slot in either foreground or background
- The foreground call is ACTIVE or IDLE or DISCONNECTED.
Change-Id: I0124d600fd8c63b8c608301f3889b3faec47f1db
Wait until all APN's have been tried before checking for permanent errors
and then, don't do retires only if all of the APN's had permanent errors.
Also, don't disable the requested apn type because if we do we won't
be able to setup data because there won't be an apn type.
This was tested by creating a new non existent APN, I chose:
Name="badapn1"
APN="badapn1"
Server="noapn.com"
Then selecting "badapn1" will cause a permanent error.
bug: 3202729
Change-Id: I182c7197456c849176ce08d7d1459359f8c3b30e
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
The problem was that when we did a contact lookup based on a SIP address,
the resulting CallerInfo object did not have the person_id field set
correctly. That meant we had no way to look up the photo for that person.
This was because of a missing case in the logic to determine which column
(in the resulting cursor) to use for the person_id lookup. We were
handling lookups fine in the PhoneLookup and Phone tables, but were
missing a case for direct lookups in the Data table (which is how we look
up SIP addresses.)
The fix is to add a case for URIs like
"content://com.android.contacts/data" when looking up the person_id.
Also, since the person_id lookup is pretty hairy (and includes ~20 lines
of comments to explain what it's doing!) refactor it out into a helper
method.
TESTED: Both SIP and PSTN calls; verified that contact name *and* photo
are displayed correctly in all cases.
Bug: 3121292
Change-Id: I2b0083cc5394c1a49bbdc9a4e5651854aedb82f7
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
+ 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
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
