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* Ignore duplicate WAP push PDU source/destination port if configuredSoojung Shin2011-06-103-1/+62
| | | | | | | | | | | | | | Some carriers duplicate the OMADM wap push PDU source/destination port. e.g. MSGTYPE-TotalSegments-CurrentSegment -SourcePortDestPort-SourcePortDestPort-OMADM PDU So the client has to ignore the duplicate source/destination port. Change-Id: I83df6e8e7d2e2e4275036a1b574247f9f40c5cf4 Signed-off-by: Soojung Shin <sj46.shin@samsung.com>
* Fixes for SMS Cell Broadcast support.Jake Hamby2011-06-064-72/+234
| | | | | | | | Add support for ETWS primary notification messages. Add method for easy concatenation of GSM multi-part broadcasts. Add test cases for SmsCbHeader, SmsCbMessage and IntRangeManager. Change-Id: Ifc646a011e79ad6c7eace9afcf84b1216eb42b7a
* Enable full support for SMS Cell Broadcast.Jake Hamby2011-06-0116-98/+1869
| | | | | | | | | | | | | Implement full support for SMS Cell Broadcast (3GPP TS 23.041). Includes support for ETWS and CMAS emergency message types. Includes GSM and UMTS support (CDMA will be added later). Note: the change to GsmAlphabet.java is only necessary if the SMS national languages support patch has been applied. If that change has not been applied, then the changes to GsmAlphabet.java in this patch set can safely be ignored. Change-Id: Ia0362c53695b8ef9a0982f558f1cffa912def34b
* SmsMessage: convert + to international dialing prefixSoojung Shin2011-05-231-1/+7
| | | | | | | | | | | When sending an SMS to an international number in the format +401234567890, the "+" should be converted to the International Dialing Prefix (in the US, 011). However, the device drops this "+" altogether in the outbound data burst message causing the message to fail or be sent to the wrong address. Change-Id: If25c092d283f1703b49cf52d0379efa54639f093 Signed-off-by: Soojung Shin <sj46.shin@samsung.com>
* Enable support for SMS national language shift tables.Jake Hamby2011-04-069-410/+970
| | | | | | | | | | | | | | | | | | | | | | | | | | | | Add support for encoding and decoding SMS 7 bit user data using the national language shift tables defined in 3GPP TS 23.038 (GSM/UMTS only), including the new tables added in Release 9 for Indic languages. Decoding is always supported, but encoding is only enabled for the specific language tables added to the new integer array resources "config_sms_enabled_single_shift_tables" and "config_sms_enabled_locking_shift_tables" defined in frameworks/base/core/res/res/values/config.xml. The default empty arrays should be overridden in an OEM overlay for the specific nationalities where SMS national language shift table encoding is allowed/mandated (e.g. Turkey). GsmAlphabet.countGsmSeptets() will try to find the most efficient encoding among all combinations of enabled locking shift and single shift tables. If no 7 bit encoding is possible, 16 bit UCS-2 encoding will be used. This change also fixes a bug in the decoder: when an escape septet is followed by a septet with no entry in the extension (single shift) table, TS 23.038 Table 6.2.1.1 states that the MS shall display the character in the main GSM 7 bit default alphabet table, or the active national language locking shift table. Previously, we were decoding this sequence as a space character. Two consecutive escape septets will continue to decode as a space character, according to Note 1 of table 6.2.1.1. Change-Id: I4dab3f0ffe39f3df2064ed93c9c05f26e274d18b
* Don't accept a user-defined dun-capable connectionRobert Greenwalt2011-03-091-0/+6
| | | | | | | | | 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
* mms: mms wap push length check before pdu processing.Soojung Shin2011-03-071-9/+9
| | | | | Change-Id: Id5b5572bcd2953e496f03142d13bd2d012225e30 Signed-off-by: Soojung Shin <sj46.shin@samsung.com>
* Added to check for supporting mms content-disposition, utf8 decoding.Soojung Shin2011-03-041-13/+36
| | | | | Change-Id: Ieae1bb2ac36675f569fb21285ca6ef289c367bf7 Signed-off-by: Soojung Shin <sj46.shin@samsung.com>
* Add a protocol property to the APNs and use it.Lorenzo Colitti2011-02-284-29/+92
| | | | | | | | | | | | | | | | | | | | | | | | | | | | 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
* Make setupDataCall take a protocol parameter.Lorenzo Colitti2011-02-287-21/+38
| | | | | | | | | | 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
* Crespo4G: Voicemail: Add 'config_telephony_...' for SprintSamyeoul Choi2011-02-251-1/+8
| | | | | | | | | | | | | | '*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
* Only decode EXT1 tag when it is available.John Wang2011-02-141-1/+8
| | | | | | | | 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
* Merge from open-source gingerbreadJean-Baptiste Queru2011-02-032-2/+26
|\ | | | | | | Change-Id: I19c4ba36cf4f2ef518b55768360b0bff1a92a5ab
| * Fix the Multi-page SMS sending error to several receipentsSang-Jun Park2011-02-032-2/+26
| | | | | | | | | | Change-Id: Iefde94b638413e3c1761f17c3065b20a044e5958 Signed-off-by: Sang-Jun Park <sj2202.park@samsung.com>
* | Merge from open-source gingerbreadJean-Baptiste Queru2011-02-031-1/+6
|\ \ | |/ | | | | Change-Id: I63e8abc1b8d6db05dfce178ae736d8d0586f6c52
| * Merge "Fix delivery report error with PENDING status in SMS" into gingerbreadJean-Baptiste Queru2011-02-031-1/+6
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| | * Fix delivery report error with PENDING status in SMSSang-Jun Park2011-02-011-1/+6
| | | | | | | | | | | | | | | | | | | | | 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
* | | two digit number handling in croatia and serbiaIrfan Sheriff2011-02-021-1/+26
| | | | | | | | | | | | | | | | | | | | | | | | If users dial 92-96, dial them normally and not treat as USSD Change-Id: If3b6cb37b7ec0ff99d76cb10cba53368094a0b5d Signed-off-by: sj2202.park@samsung.com
* | | Merge from open-source gingerbreadJean-Baptiste Queru2011-02-021-0/+41
|\ \ \ | |/ / | | | | | | Change-Id: Iec6167bec8423e39dde053f23969c1c76e10a461
| * | fix for supporting 3 digits MNC codeSang-Jun Park2011-02-021-0/+41
| |/ | | | | | | | | | | | | | | 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>
* | Allow TTY mode for GSM PhonesEric Laurent2011-02-015-34/+32
| | | | | | | | | | | | | | | | | | | | | | 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
* | Add some network types that OEM's are asking for.Robert Greenwalt2011-01-281-1/+4
| | | | | | | | | | | | | | | | 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
* | do not merge bug 3370834 Cherrypick from masterJean-Michel Trivi2011-01-261-5/+10
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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
* | Merge "Get mute state from active call." into gingerbreadHung-ying Tyan2011-01-242-1/+5
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| * | Get mute state from active call.Hung-ying Tyan2011-01-112-1/+5
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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
* | | Enable recovery in RIL wakelock release check.John Wang2011-01-211-18/+56
|/ / | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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
* | Merge "Fix setting audio group mode in SipPhone." into gingerbreadHung-ying Tyan2011-01-041-21/+50
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| * | Fix setting audio group mode in SipPhone.Hung-ying Tyan2010-12-291-21/+50
| | | | | | | | | | | | | | | Bug: 3119690 Change-Id: I495d3c031ee4c272d360fe19553ef9726a3f8771
* | | Clear request list while timeout.John Wang2010-12-281-24/+47
|/ / | | | | | | | | | | | | | | | | | | | | 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
* | Fix policy issues when screen is off. (DO NOT MERGE)Jeff Brown2010-12-221-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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
* | Fix SIP bug of different transport/port used for requests.Chung-yih Wang2010-12-071-1/+3
| | | | | | | | | | bug: http://b/3156148 Change-Id: I4fa5b274d2e90ebde12d9e99822dc193a65bad32
* | Add "canDial" check.John Wang2010-12-011-0/+30
| | | | | | | | | | | | | | | | | | | | | | | | | | 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
* | Merge "Do not suppress error feedback during a SIP call." into gingerbreadHung-ying Tyan2010-11-301-6/+0
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| * | Do not suppress error feedback during a SIP call.Hung-ying Tyan2010-11-301-6/+0
| | | | | | | | | | | | | | | Bug: 3124788 Change-Id: Ia0a06f72336d1795515428eba0c9f875c32d13d1
* | | Reduce the outrageous verbosity of CallerInfo.toString().David Brown2010-11-301-22/+35
| | | | | | | | | | | | | | | Bug: 3121292 Change-Id: Ia8383891ef29a003acbd627b25ce87a187ef95c0
* | | Merge "Fix bug 3121292: Contact photo not shown correctly for SIP calls" ↵David Brown2010-11-301-28/+90
|\ \ \ | | | | | | | | | | | | into gingerbread
| * | | Fix bug 3121292: Contact photo not shown correctly for SIP callsDavid Brown2010-10-261-28/+90
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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
* | | | Merge "Fix GSM permanent failure handling, DO NOT MERGE." into gingerbreadWink Saville2010-11-301-12/+18
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| * | | Fix GSM permanent failure handling, DO NOT MERGE.Wink Saville2010-11-171-12/+18
| | |/ | |/| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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
* | | Merge "Throw proper exceptions in SipManager" into gingerbreadHung-ying Tyan2010-11-301-2/+1
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| * | Throw proper exceptions in SipManagerHung-ying Tyan2010-11-031-2/+1
| |/ | | | | | | | | | | | | | | | | | | | | | | | | | | 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
* | Fix the audio mode glitch during hangup.John Wang2010-11-101-1/+5
|/ | | | | | | | | | 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
* Partial fix for issue 3124895.Eric Laurent2010-10-251-1/+1
| | | | | | | 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
* Fix connect duration for un-established SIP calls.Hung-ying Tyan2010-10-211-3/+3
| | | | | Bug: 3118364 Change-Id: I931b675de04a3aac70b45d6bae27ab42a84f2d1e
* Fix n-way conf call in SipPhone.Hung-ying Tyan2010-10-211-3/+20
| | | | | | | | | + 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
* Fix the build.Jeff Hamilton2010-10-201-1/+1
| | | | Change-Id: Id5bfa0f91e6ec687201a320a1eb4d8a46050875e
* Do another contact lookup if the first one fails and...Hung-ying Tyan2010-10-212-0/+34
| | | | | | | | | 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
* Silently reject a ringing call when another call is dialing/ringing.Hung-ying Tyan2010-10-211-2/+20
| | | | | | | http://b/issue?id=3109483 http://b/issue?id=3103072 Change-Id: I34f13225319c7f2a41e1ea9e25811866432ab809
* Remove ringtone API from SipAudioCall.Hung-ying Tyan2010-10-201-1/+0
| | | | | | | | (watch out auto-merge conflict for SipAudioCall). Bug: 3113033, related CL: https://android-git/g/#change,75185 Change-Id: Ib48d3b990e229e0b341e47e10e76934e1a50d10f
* Merge "Return display name in SipConnection.getCnapName()." into gingerbreadHung-ying Tyan2010-10-191-0/+13
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